Network Working Group Internet Engineering Task Force
Request for Comments: 1122 R. Braden, Editor
October 1989
Requirements for Internet Hosts -- Communication Layers
Status of This Memo
This RFC is an official specification for the Internet community. It
incorporates by reference, amends, corrects, and supplements the
primary protocol standards documents relating to hosts. Distribution
of this document is unlimited.
Summary
This is one RFC of a pair that defines and discusses the requirements
for Internet host software. This RFC covers the communications
protocol layers: link layer, IP layer, and transport layer; its
companion RFC-1123 covers the application and support protocols.
Table of Contents
1. INTRODUCTION ............................................... 5
1.1 The Internet Architecture .............................. 6
1.1.1 Internet Hosts .................................... 6
1.1.2 Architectural Assumptions ......................... 7
1.1.3 Internet Protocol Suite ........................... 8
1.1.4 Embedded Gateway Code ............................. 10
1.2 General Considerations ................................. 12
1.2.1 Continuing Internet Evolution ..................... 12
1.2.2 Robustness Principle .............................. 12
1.2.3 Error Logging ..................................... 13
1.2.4 Configuration ..................................... 14
1.3 Reading this Document .................................. 15
1.3.1 Organization ...................................... 15
1.3.2 Requirements ...................................... 16
1.3.3 Terminology ....................................... 17
1.4 Acknowledgments ........................................ 20
2. LINK LAYER .................................................. 21
2.1 INTRODUCTION ........................................... 21
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2.2 PROTOCOL WALK-THROUGH .................................. 21
2.3 SPECIFIC ISSUES ........................................ 21
2.3.1 Trailer Protocol Negotiation ...................... 21
2.3.2 Address Resolution Protocol -- ARP ................ 22
2.3.2.1 ARP Cache Validation ......................... 22
2.3.2.2 ARP Packet Queue ............................. 24
2.3.3 Ethernet and IEEE 802 Encapsulation ............... 24
2.4 LINK/INTERNET LAYER INTERFACE .......................... 25
2.5 LINK LAYER REQUIREMENTS SUMMARY ........................ 26
3. INTERNET LAYER PROTOCOLS .................................... 27
3.1 INTRODUCTION ............................................ 27
3.2 PROTOCOL WALK-THROUGH .................................. 29
3.2.1 Internet Protocol -- IP ............................ 29
3.2.1.1 Version Number ............................... 29
3.2.1.2 Checksum ..................................... 29
3.2.1.3 Addressing ................................... 29
3.2.1.4 Fragmentation and Reassembly ................. 32
3.2.1.5 Identification ............................... 32
3.2.1.6 Type-of-Service .............................. 33
3.2.1.7 Time-to-Live ................................. 34
3.2.1.8 Options ...................................... 35
3.2.2 Internet Control Message Protocol -- ICMP .......... 38
3.2.2.1 Destination Unreachable ...................... 39
3.2.2.2 Redirect ..................................... 40
3.2.2.3 Source Quench ................................ 41
3.2.2.4 Time Exceeded ................................ 41
3.2.2.5 Parameter Problem ............................ 42
3.2.2.6 Echo Request/Reply ........................... 42
3.2.2.7 Information Request/Reply .................... 43
3.2.2.8 Timestamp and Timestamp Reply ................ 43
3.2.2.9 Address Mask Request/Reply ................... 45
3.2.3 Internet Group Management Protocol IGMP ........... 47
3.3 SPECIFIC ISSUES ........................................ 47
3.3.1 Routing Outbound Datagrams ........................ 47
3.3.1.1 Local/Remote Decision ........................ 47
3.3.1.2 Gateway Selection ............................ 48
3.3.1.3 Route Cache .................................. 49
3.3.1.4 Dead Gateway Detection ....................... 51
3.3.1.5 New Gateway Selection ........................ 55
3.3.1.6 Initialization ............................... 56
3.3.2 Reassembly ........................................ 56
3.3.3 Fragmentation ..................................... 58
3.3.4 Local Multihoming ................................. 60
3.3.4.1 Introduction ................................. 60
3.3.4.2 Multihoming Requirements ..................... 61
3.3.4.3 Choosing a Source Address .................... 64
3.3.5 Source Route Forwarding ........................... 65
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3.3.6 Broadcasts ........................................ 66
3.3.7 IP Multicasting ................................... 67
3.3.8 Error Reporting ................................... 69
3.4 INTERNET/TRANSPORT LAYER INTERFACE ..................... 69
3.5 INTERNET LAYER REQUIREMENTS SUMMARY .................... 72
4. TRANSPORT PROTOCOLS ......................................... 77
4.1 USER DATAGRAM PROTOCOL -- UDP .......................... 77
4.1.1 INTRODUCTION ...................................... 77
4.1.2 PROTOCOL WALK-THROUGH ............................. 77
4.1.3 SPECIFIC ISSUES ................................... 77
4.1.3.1 Ports ........................................ 77
4.1.3.2 IP Options ................................... 77
4.1.3.3 ICMP Messages ................................ 78
4.1.3.4 UDP Checksums ................................ 78
4.1.3.5 UDP Multihoming .............................. 79
4.1.3.6 Invalid Addresses ............................ 79
4.1.4 UDP/APPLICATION LAYER INTERFACE ................... 79
4.1.5 UDP REQUIREMENTS SUMMARY .......................... 80
4.2 TRANSMISSION CONTROL PROTOCOL -- TCP ................... 82
4.2.1 INTRODUCTION ...................................... 82
4.2.2 PROTOCOL WALK-THROUGH ............................. 82
4.2.2.1 Well-Known Ports ............................. 82
4.2.2.2 Use of Push .................................. 82
4.2.2.3 Window Size .................................. 83
4.2.2.4 Urgent Pointer ............................... 84
4.2.2.5 TCP Options .................................. 85
4.2.2.6 Maximum Segment Size Option .................. 85
4.2.2.7 TCP Checksum ................................. 86
4.2.2.8 TCP Connection State Diagram ................. 86
4.2.2.9 Initial Sequence Number Selection ............ 87
4.2.2.10 Simultaneous Open Attempts .................. 87
4.2.2.11 Recovery from Old Duplicate SYN ............. 87
4.2.2.12 RST Segment ................................. 87
4.2.2.13 Closing a Connection ........................ 87
4.2.2.14 Data Communication .......................... 89
4.2.2.15 Retransmission Timeout ...................... 90
4.2.2.16 Managing the Window ......................... 91
4.2.2.17 Probing Zero Windows ........................ 92
4.2.2.18 Passive OPEN Calls .......................... 92
4.2.2.19 Time to Live ................................ 93
4.2.2.20 Event Processing ............................ 93
4.2.2.21 Acknowledging Queued Segments ............... 94
4.2.3 SPECIFIC ISSUES ................................... 95
4.2.3.1 Retransmission Timeout Calculation ........... 95
4.2.3.2 When to Send an ACK Segment .................. 96
4.2.3.3 When to Send a Window Update ................. 97
4.2.3.4 When to Send Data ............................ 98
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4.2.3.5 TCP Connection Failures ...................... 100
4.2.3.6 TCP Keep-Alives .............................. 101
4.2.3.7 TCP Multihoming .............................. 103
4.2.3.8 IP Options ................................... 103
4.2.3.9 ICMP Messages ................................ 103
4.2.3.10 Remote Address Validation ................... 104
4.2.3.11 TCP Traffic Patterns ........................ 104
4.2.3.12 Efficiency .................................. 105
4.2.4 TCP/APPLICATION LAYER INTERFACE ................... 106
4.2.4.1 Asynchronous Reports ......................... 106
4.2.4.2 Type-of-Service .............................. 107
4.2.4.3 Flush Call ................................... 107
4.2.4.4 Multihoming .................................. 108
4.2.5 TCP REQUIREMENT SUMMARY ........................... 108
5. REFERENCES ................................................. 112
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1. INTRODUCTION
This document is one of a pair that defines and discusses the
requirements for host system implementations of the Internet protocol
suite. This RFC covers the communication protocol layers: link
layer, IP layer, and transport layer. Its companion RFC,
"Requirements for Internet Hosts -- Application and Support"
[INTRO:1], covers the application layer protocols. This document
should also be read in conjunction with "Requirements for Internet
Gateways" [INTRO:2].
These documents are intended to provide guidance for vendors,
implementors, and users of Internet communication software. They
represent the consensus of a large body of technical experience and
wisdom, contributed by the members of the Internet research and
vendor communities.
This RFC enumerates standard protocols that a host connected to the
Internet must use, and it incorporates by reference the RFCs and
other documents describing the current specifications for these
protocols. It corrects errors in the referenced documents and adds
additional discussion and guidance for an implementor.
For each protocol, this document also contains an explicit set of
requirements, recommendations, and options. The reader must
understand that the list of requirements in this document is
incomplete by itself; the complete set of requirements for an
Internet host is primarily defined in the standard protocol
specification documents, with the corrections, amendments, and
supplements contained in this RFC.
A good-faith implementation of the protocols that was produced after
careful reading of the RFC's and with some interaction with the
Internet technical community, and that followed good communications
software engineering practices, should differ from the requirements
of this document in only minor ways. Thus, in many cases, the
"requirements" in this RFC are already stated or implied in the
standard protocol documents, so that their inclusion here is, in a
sense, redundant. However, they were included because some past
implementation has made the wrong choice, causing problems of
interoperability, performance, and/or robustness.
This document includes discussion and explanation of many of the
requirements and recommendations. A simple list of requirements
would be dangerous, because:
o Some required features are more important than others, and some
features are optional.
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o There may be valid reasons why particular vendor products that
are designed for restricted contexts might choose to use
different specifications.
However, the specifications of this document must be followed to meet
the general goal of arbitrary host interoperation across the
diversity and complexity of the Internet system. Although most
current implementations fail to meet these requirements in various
ways, some minor and some major, this specification is the ideal
towards which we need to move.
These requirements are based on the current level of Internet
architecture. This document will be updated as required to provide
additional clarifications or to include additional information in
those areas in which specifications are still evolving.
This introductory section begins with a brief overview of the
Internet architecture as it relates to hosts, and then gives some
general advice to host software vendors. Finally, there is some
guidance on reading the rest of the document and some terminology.
1.1 The Internet Architecture
General background and discussion on the Internet architecture and
supporting protocol suite can be found in the DDN Protocol
Handbook [INTRO:3]; for background see for example [INTRO:9],
[INTRO:10], and [INTRO:11]. Reference [INTRO:5] describes the
procedure for obtaining Internet protocol documents, while
[INTRO:6] contains a list of the numbers assigned within Internet
protocols.
1.1.1 Internet Hosts
A host computer, or simply "host," is the ultimate consumer of
communication services. A host generally executes application
programs on behalf of user(s), employing network and/or
Internet communication services in support of this function.
An Internet host corresponds to the concept of an "End-System"
used in the OSI protocol suite [INTRO:13].
An Internet communication system consists of interconnected
packet networks supporting communication among host computers
using the Internet protocols. The networks are interconnected
using packet-switching computers called "gateways" or "IP
routers" by the Internet community, and "Intermediate Systems"
by the OSI world [INTRO:13]. The RFC "Requirements for
Internet Gateways" [INTRO:2] contains the official
specifications for Internet gateways. That RFC together with
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the present document and its companion [INTRO:1] define the
rules for the current realization of the Internet architecture.
Internet hosts span a wide range of size, speed, and function.
They range in size from small microprocessors through
workstations to mainframes and supercomputers. In function,
they range from single-purpose hosts (such as terminal servers)
to full-service hosts that support a variety of online network
services, typically including remote login, file transfer, and
electronic mail.
A host is generally said to be multihomed if it has more than
one interface to the same or to different networks. See
Section 1.1.3 on "Terminology".
1.1.2 Architectural Assumptions
The current Internet architecture is based on a set of
assumptions about the communication system. The assumptions
most relevant to hosts are as follows:
(a) The Internet is a network of networks.
Each host is directly connected to some particular
network(s); its connection to the Internet is only
conceptual. Two hosts on the same network communicate
with each other using the same set of protocols that they
would use to communicate with hosts on distant networks.
(b) Gateways don't keep connection state information.
To improve robustness of the communication system,
gateways are designed to be stateless, forwarding each IP
datagram independently of other datagrams. As a result,
redundant paths can be exploited to provide robust service
in spite of failures of intervening gateways and networks.
All state information required for end-to-end flow control
and reliability is implemented in the hosts, in the
transport layer or in application programs. All
connection control information is thus co-located with the
end points of the communication, so it will be lost only
if an end point fails.
(c) Routing complexity should be in the gateways.
Routing is a complex and difficult problem, and ought to
be performed by the gateways, not the hosts. An important
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objective is to insulate host software from changes caused
by the inevitable evolution of the Internet routing
architecture.
(d) The System must tolerate wide network variation.
A basic objective of the Internet design is to tolerate a
wide range of network characteristics -- e.g., bandwidth,
delay, packet loss, packet reordering, and maximum packet
size. Another objective is robustness against failure of
individual networks, gateways, and hosts, using whatever
bandwidth is still available. Finally, the goal is full
"open system interconnection": an Internet host must be
able to interoperate robustly and effectively with any
other Internet host, across diverse Internet paths.
Sometimes host implementors have designed for less
ambitious goals. For example, the LAN environment is
typically much more benign than the Internet as a whole;
LANs have low packet loss and delay and do not reorder
packets. Some vendors have fielded host implementations
that are adequate for a simple LAN environment, but work
badly for general interoperation. The vendor justifies
such a product as being economical within the restricted
LAN market. However, isolated LANs seldom stay isolated
for long; they are soon gatewayed to each other, to
organization-wide internets, and eventually to the global
Internet system. In the end, neither the customer nor the
vendor is served by incomplete or substandard Internet
host software.
The requirements spelled out in this document are designed
for a full-function Internet host, capable of full
interoperation over an arbitrary Internet path.
1.1.3 Internet Protocol Suite
To communicate using the Internet system, a host must implement
the layered set of protocols comprising the Internet protocol
suite. A host typically must implement at least one protocol
from each layer.
The protocol layers used in the Internet architecture are as
follows [INTRO:4]:
o Application Layer
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The application layer is the top layer of the Internet
protocol suite. The Internet suite does not further
subdivide the application layer, although some of the
Internet application layer protocols do contain some
internal sub-layering. The application layer of the
Internet suite essentially combines the functions of the
top two layers -- Presentation and Application -- of the
OSI reference model.
We distinguish two categories of application layer
protocols: user protocols that provide service directly
to users, and support protocols that provide common system
functions. Requirements for user and support protocols
will be found in the companion RFC [INTRO:1].
The most common Internet user protocols are:
o Telnet (remote login)
o FTP (file transfer)
o SMTP (electronic mail delivery)
There are a number of other standardized user protocols
[INTRO:4] and many private user protocols.
Support protocols, used for host name mapping, booting,
and management, include SNMP, BOOTP, RARP, and the Domain
Name System (DNS) protocols.
o Transport Layer
The transport layer provides end-to-end communication
services for applications. There are two primary
transport layer protocols at present:
o Transmission Control Protocol (TCP)
o User Datagram Protocol (UDP)
TCP is a reliable connection-oriented transport service
that provides end-to-end reliability, resequencing, and
flow control. UDP is a connectionless ("datagram")
transport service.
Other transport protocols have been developed by the
research community, and the set of official Internet
transport protocols may be expanded in the future.
Transport layer protocols are discussed in Chapter 4.
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o Internet Layer
All Internet transport protocols use the Internet Protocol
(IP) to carry data from source host to destination host.
IP is a connectionless or datagram internetwork service,
providing no end-to-end delivery guarantees. Thus, IP
datagrams may arrive at the destination host damaged,
duplicated, out of order, or not at all. The layers above
IP are responsible for reliable delivery service when it
is required. The IP protocol includes provision for
addressing, type-of-service specification, fragmentation
and reassembly, and security information.
The datagram or connectionless nature of the IP protocol
is a fundamental and characteristic feature of the
Internet architecture. Internet IP was the model for the
OSI Connectionless Network Protocol [INTRO:12].
ICMP is a control protocol that is considered to be an
integral part of IP, although it is architecturally
layered upon IP, i.e., it uses IP to carry its data end-
to-end just as a transport protocol like TCP or UDP does.
ICMP provides error reporting, congestion reporting, and
first-hop gateway redirection.
IGMP is an Internet layer protocol used for establishing
dynamic host groups for IP multicasting.
The Internet layer protocols IP, ICMP, and IGMP are
discussed in Chapter 3.
o Link Layer
To communicate on its directly-connected network, a host
must implement the communication protocol used to
interface to that network. We call this a link layer or
media-access layer protocol.
There is a wide variety of link layer protocols,
corresponding to the many different types of networks.
See Chapter 2.
1.1.4 Embedded Gateway Code
Some Internet host software includes embedded gateway
functionality, so that these hosts can forward packets as a
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gateway would, while still performing the application layer
functions of a host.
Such dual-purpose systems must follow the Gateway Requirements
RFC [INTRO:2] with respect to their gateway functions, and
must follow the present document with respect to their host
functions. In all overlapping cases, the two specifications
should be in agreement.
There are varying opinions in the Internet community about
embedded gateway functionality. The main arguments are as
follows:
o Pro: in a local network environment where networking is
informal, or in isolated internets, it may be convenient
and economical to use existing host systems as gateways.
There is also an architectural argument for embedded
gateway functionality: multihoming is much more common
than originally foreseen, and multihoming forces a host to
make routing decisions as if it were a gateway. If the
multihomed host contains an embedded gateway, it will
have full routing knowledge and as a result will be able
to make more optimal routing decisions.
o Con: Gateway algorithms and protocols are still changing,
and they will continue to change as the Internet system
grows larger. Attempting to include a general gateway
function within the host IP layer will force host system
maintainers to track these (more frequent) changes. Also,
a larger pool of gateway implementations will make
coordinating the changes more difficult. Finally, the
complexity of a gateway IP layer is somewhat greater than
that of a host, making the implementation and operation
tasks more complex.
In addition, the style of operation of some hosts is not
appropriate for providing stable and robust gateway
service.
There is considerable merit in both of these viewpoints. One
conclusion can be drawn: an host administrator must have
conscious control over whether or not a given host acts as a
gateway. See Section 3.1 for the detailed requirements.
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1.2 General Considerations
There are two important lessons that vendors of Internet host
software have learned and which a new vendor should consider
seriously.
1.2.1 Continuing Internet Evolution
The enormous growth of the Internet has revealed problems of
management and scaling in a large datagram-based packet
communication system. These problems are being addressed, and
as a result there will be continuing evolution of the
specifications described in this document. These changes will
be carefully planned and controlled, since there is extensive
participation in this planning by the vendors and by the
organizations responsible for operations of the networks.
Development, evolution, and revision are characteristic of
computer network protocols today, and this situation will
persist for some years. A vendor who develops computer
communication software for the Internet protocol suite (or any
other protocol suite!) and then fails to maintain and update
that software for changing specifications is going to leave a
trail of unhappy customers. The Internet is a large
communication network, and the users are in constant contact
through it. Experience has shown that knowledge of
deficiencies in vendor software propagates quickly through the
Internet technical community.
1.2.2 Robustness Principle
At every layer of the protocols, there is a general rule whose
application can lead to enormous benefits in robustness and
interoperability [IP:1]:
"Be liberal in what you accept, and
conservative in what you send"
Software should be written to deal with every conceivable
error, no matter how unlikely; sooner or later a packet will
come in with that particular combination of errors and
attributes, and unless the software is prepared, chaos can
ensue. In general, it is best to assume that the network is
filled with malevolent entities that will send in packets
designed to have the worst possible effect. This assumption
will lead to suitable protective design, although the most
serious problems in the Internet have been caused by
unenvisaged mechanisms triggered by low-probability events;
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mere human malice would never have taken so devious a course!
Adaptability to change must be designed into all levels of
Internet host software. As a simple example, consider a
protocol specification that contains an enumeration of values
for a particular header field -- e.g., a type field, a port
number, or an error code; this enumeration must be assumed to
be incomplete. Thus, if a protocol specification defines four
possible error codes, the software must not break when a fifth
code shows up. An undefined code might be logged (see below),
but it must not cause a failure.
The second part of the principle is almost as important:
software on other hosts may contain deficiencies that make it
unwise to exploit legal but obscure protocol features. It is
unwise to stray far from the obvious and simple, lest untoward
effects result elsewhere. A corollary of this is "watch out
for misbehaving hosts"; host software should be prepared, not
just to survive other misbehaving hosts, but also to cooperate
to limit the amount of disruption such hosts can cause to the
shared communication facility.
1.2.3 Error Logging
The Internet includes a great variety of host and gateway
systems, each implementing many protocols and protocol layers,
and some of these contain bugs and mis-features in their
Internet protocol software. As a result of complexity,
diversity, and distribution of function, the diagnosis of
Internet problems is often very difficult.
Problem diagnosis will be aided if host implementations include
a carefully designed facility for logging erroneous or
"strange" protocol events. It is important to include as much
diagnostic information as possible when an error is logged. In
particular, it is often useful to record the header(s) of a
packet that caused an error. However, care must be taken to
ensure that error logging does not consume prohibitive amounts
of resources or otherwise interfere with the operation of the
host.
There is a tendency for abnormal but harmless protocol events
to overflow error logging files; this can be avoided by using a
"circular" log, or by enabling logging only while diagnosing a
known failure. It may be useful to filter and count duplicate
successive messages. One strategy that seems to work well is:
(1) always count abnormalities and make such counts accessible
through the management protocol (see [INTRO:1]); and (2) allow
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the logging of a great variety of events to be selectively
enabled. For example, it might useful to be able to "log
everything" or to "log everything for host X".
Note that different managements may have differing policies
about the amount of error logging that they want normally
enabled in a host. Some will say, "if it doesn't hurt me, I
don't want to know about it", while others will want to take a
more watchful and aggressive attitude about detecting and
removing protocol abnormalities.
1.2.4 Configuration
It would be ideal if a host implementation of the Internet
protocol suite could be entirely self-configuring. This would
allow the whole suite to be implemented in ROM or cast into
silicon, it would simplify diskless workstations, and it would
be an immense boon to harried LAN administrators as well as
system vendors. We have not reached this ideal; in fact, we
are not even close.
At many points in this document, you will find a requirement
that a parameter be a configurable option. There are several
different reasons behind such requirements. In a few cases,
there is current uncertainty or disagreement about the best
value, and it may be necessary to update the recommended value
in the future. In other cases, the value really depends on
external factors -- e.g., the size of the host and the
distribution of its communication load, or the speeds and
topology of nearby networks -- and self-tuning algorithms are
unavailable and may be insufficient. In some cases,
configurability is needed because of administrative
requirements.
Finally, some configuration options are required to communicate
with obsolete or incorrect implementations of the protocols,
distributed without sources, that unfortunately persist in many
parts of the Internet. To make correct systems coexist with
these faulty systems, administrators often have to "mis-
configure" the correct systems. This problem will correct
itself gradually as the faulty systems are retired, but it
cannot be ignored by vendors.
When we say that a parameter must be configurable, we do not
intend to require that its value be explicitly read from a
configuration file at every boot time. We recommend that
implementors set up a default for each parameter, so a
configuration file is only necessary to override those defaults
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that are inappropriate in a particular installation. Thus, the
configurability requirement is an assurance that it will be
POSSIBLE to override the default when necessary, even in a
binary-only or ROM-based product.
This document requires a particular value for such defaults in
some cases. The choice of default is a sensitive issue when
the configuration item controls the accommodation to existing
faulty systems. If the Internet is to converge successfully to
complete interoperability, the default values built into
implementations must implement the official protocol, not
"mis-configurations" to accommodate faulty implementations.
Although marketing considerations have led some vendors to
choose mis-configuration defaults, we urge vendors to choose
defaults that will conform to the standard.
Finally, we note that a vendor needs to provide adequate
documentation on all configuration parameters, their limits and
effects.
1.3 Reading this Document
1.3.1 Organization
Protocol layering, which is generally used as an organizing
principle in implementing network software, has also been used
to organize this document. In describing the rules, we assume
that an implementation does strictly mirror the layering of the
protocols. Thus, the following three major sections specify
the requirements for the link layer, the internet layer, and
the transport layer, respectively. A companion RFC [INTRO:1]
covers application level software. This layerist organization
was chosen for simplicity and clarity.
However, strict layering is an imperfect model, both for the
protocol suite and for recommended implementation approaches.
Protocols in different layers interact in complex and sometimes
subtle ways, and particular functions often involve multiple
layers. There are many design choices in an implementation,
many of which involve creative "breaking" of strict layering.
Every implementor is urged to read references [INTRO:7] and
[INTRO:8].
This document describes the conceptual service interface
between layers using a functional ("procedure call") notation,
like that used in the TCP specification [TCP:1]. A host
implementation must support the logical information flow
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implied by these calls, but need not literally implement the
calls themselves. For example, many implementations reflect
the coupling between the transport layer and the IP layer by
giving them shared access to common data structures. These
data structures, rather than explicit procedure calls, are then
the agency for passing much of the information that is
required.
In general, each major section of this document is organized
into the following subsections:
(1) Introduction
(2) Protocol Walk-Through -- considers the protocol
specification documents section-by-section, correcting
errors, stating requirements that may be ambiguous or
ill-defined, and providing further clarification or
explanation.
(3) Specific Issues -- discusses protocol design and
implementation issues that were not included in the walk-
through.
(4) Interfaces -- discusses the service interface to the next
higher layer.
(5) Summary -- contains a summary of the requirements of the
section.
Under many of the individual topics in this document, there is
parenthetical material labeled "DISCUSSION" or
"IMPLEMENTATION". This material is intended to give
clarification and explanation of the preceding requirements
text. It also includes some suggestions on possible future
directions or developments. The implementation material
contains suggested approaches that an implementor may want to
consider.
The summary sections are intended to be guides and indexes to
the text, but are necessarily cryptic and incomplete. The
summaries should never be used or referenced separately from
the complete RFC.
1.3.2 Requirements
In this document, the words that are used to define the
significance of each particular requirement are capitalized.
Internet Engineering Task Force [Page 16]
RFC1122 INTRODUCTION October 1989
These words are:
* "MUST"
This word or the adjective "REQUIRED" means that the item
is an absolute requirement of the specification.
* "SHOULD"
This word or the adjective "RECOMMENDED" means that there
may exist valid reasons in particular circumstances to
ignore this item, but the full implications should be
understood and the case carefully weighed before choosing
a different course.
* "MAY"
This word or the adjective "OPTIONAL" means that this item
is truly optional. One vendor may choose to include the
item because a particular marketplace requires it or
because it enhances the product, for example; another
vendor may omit the same item.
An implementation is not compliant if it fails to satisfy one
or more of the MUST requirements for the protocols it
implements. An implementation that satisfies all the MUST and
all the SHOULD requirements for its protocols is said to be
"unconditionally compliant"; one that satisfies all the MUST
requirements but not all the SHOULD requirements for its
protocols is said to be "conditionally compliant".
1.3.3 Terminology
This document uses the following technical terms:
Segment
A segment is the unit of end-to-end transmission in the
TCP protocol. A segment consists of a TCP header followed
by application data. A segment is transmitted by
encapsulation inside an IP datagram.
Message
In this description of the lower-layer protocols, a
message is the unit of transmission in a transport layer
protocol. In particular, a TCP segment is a message. A
message consists of a transport protocol header followed
by application protocol data. To be transmitted end-to-
Internet Engineering Task Force [Page 17]
RFC1122 INTRODUCTION October 1989
end through the Internet, a message must be encapsulated
inside a datagram.
IP Datagram
An IP datagram is the unit of end-to-end transmission in
the IP protocol. An IP datagram consists of an IP header
followed by transport layer data, i.e., of an IP header
followed by a message.
In the description of the internet layer (Section 3), the
unqualified term "datagram" should be understood to refer
to an IP datagram.
Packet
A packet is the unit of data passed across the interface
between the internet layer and the link layer. It
includes an IP header and data. A packet may be a
complete IP datagram or a fragment of an IP datagram.
Frame
A frame is the unit of transmission in a link layer
protocol, and consists of a link-layer header followed by
a packet.
Connected Network
A network to which a host is interfaced is often known as
the "local network" or the "subnetwork" relative to that
host. However, these terms can cause confusion, and
therefore we use the term "connected network" in this
document.
Multihomed
A host is said to be multihomed if it has multiple IP
addresses. For a discussion of multihoming, see Section
3.3.4 below.
Physical network interface
This is a physical interface to a connected network and
has a (possibly unique) link-layer address. Multiple
physical network interfaces on a single host may share the
same link-layer address, but the address must be unique
for different hosts on the same physical network.
Logical [network] interface
We define a logical [network] interface to be a logical
path, distinguished by a unique IP address, to a connected
network. See Section 3.3.4.
Internet Engineering Task Force [Page 18]
RFC1122 INTRODUCTION October 1989
Specific-destination address
This is the effective destination address of a datagram,
even if it is broadcast or multicast; see Section 3.2.1.3.
Path
At a given moment, all the IP datagrams from a particular
source host to a particular destination host will
typically traverse the same sequence of gateways. We use
the term "path" for this sequence. Note that a path is
uni-directional; it is not unusual to have different paths
in the two directions between a given host pair.
MTU
The maximum transmission unit, i.e., the size of the
largest packet that can be transmitted.
The terms frame, packet, datagram, message, and segment are
illustrated by the following schematic diagrams:
A. Transmission on connected network:
_______________________________________________
| LL hdr | IP hdr | (data) |
|________|________|_____________________________|
B. Before IP fragmentation or after IP reassembly:
______________________________________
| IP hdr | transport| Application Data |
|________|____hdr___|__________________|
or, for TCP:
______________________________________
| IP hdr | TCP hdr | Application Data |
|________|__________|__________________|
Internet Engineering Task Force [Page 19]
RFC1122 INTRODUCTION October 1989
1.4 Acknowledgments
This document incorporates contributions and comments from a large
group of Internet protocol experts, including representatives of
university and research labs, vendors, and government agencies.
It was assembled primarily by the Host Requirements Working Group
of the Internet Engineering Task Force (IETF).
The Editor would especially like to acknowledge the tireless
dedication of the following people, who attended many long
meetings and generated 3 million bytes of electronic mail over the
past 18 months in pursuit of this document: Philip Almquist, Dave
Borman (Cray Research), Noel Chiappa, Dave Crocker (DEC), Steve
Deering (Stanford), Mike Karels (Berkeley), Phil Karn (Bellcore),
John Lekashman (NASA), Charles Lynn (BBN), Keith McCloghrie (TWG),
Paul Mockapetris (ISI), Thomas Narten (Purdue), Craig Partridge
(BBN), Drew Perkins (CMU), and James Van Bokkelen (FTP Software).
In addition, the following people made major contributions to the
effort: Bill Barns (Mitre), Steve Bellovin (AT&T), Mike Brescia
(BBN), Ed Cain (DCA), Annette DeSchon (ISI), Martin Gross (DCA),
Phill Gross (NRI), Charles Hedrick (Rutgers), Van Jacobson (LBL),
John Klensin (MIT), Mark Lottor (SRI), Milo Medin (NASA), Bill
Melohn (Sun Microsystems), Greg Minshall (Kinetics), Jeff Mogul
(DEC), John Mullen (CMC), Jon Postel (ISI), John Romkey (Epilogue
Technology), and Mike StJohns (DCA). The following also made
significant contributions to particular areas: Eric Allman
(Berkeley), Rob Austein (MIT), Art Berggreen (ACC), Keith Bostic
(Berkeley), Vint Cerf (NRI), Wayne Hathaway (NASA), Matt Korn
(IBM), Erik Naggum (Naggum Software, Norway), Robert Ullmann
(Prime Computer), David Waitzman (BBN), Frank Wancho (USA), Arun
Welch (Ohio State), Bill Westfield (Cisco), and Rayan Zachariassen
(Toronto).
We are grateful to all, including any contributors who may have
been inadvertently omitted from this list.
Internet Engineering Task Force [Page 20]
RFC1122 LINK LAYER October 1989
2. LINK LAYER
2.1 INTRODUCTION
All Internet systems, both hosts and gateways, have the same
requirements for link layer protocols. These requirements are
given in Chapter 3 of "Requirements for Internet Gateways"
[INTRO:2], augmented with the material in this section.
2.2 PROTOCOL WALK-THROUGH
None.
2.3 SPECIFIC ISSUES
2.3.1 Trailer Protocol Negotiation
The trailer protocol [LINK:1] for link-layer encapsulation MAY
be used, but only when it has been verified that both systems
(host or gateway) involved in the link-layer communication
implement trailers. If the system does not dynamically
negotiate use of the trailer protocol on a per-destination
basis, the default configuration MUST disable the protocol.
DISCUSSION:
The trailer protocol is a link-layer encapsulation
technique that rearranges the data contents of packets
sent on the physical network. In some cases, trailers
improve the throughput of higher layer protocols by
reducing the amount of data copying within the operating
system. Higher layer protocols are unaware of trailer
use, but both the sending and receiving host MUST
understand the protocol if it is used.
Improper use of trailers can result in very confusing
symptoms. Only packets with specific size attributes are
encapsulated using trailers, and typically only a small
fraction of the packets being exchanged have these
attributes. Thus, if a system using trailers exchanges
packets with a system that does not, some packets
disappear into a black hole while others are delivered
successfully.
IMPLEMENTATION:
On an Ethernet, packets encapsulated with trailers use a
distinct Ethernet type [LINK:1], and trailer negotiation
is performed at the time that ARP is used to discover the
link-layer address of a destination system.
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RFC1122 LINK LAYER October 1989
Specifically, the ARP exchange is completed in the usual
manner using the normal IP protocol type, but a host that
wants to speak trailers will send an additional "trailer
ARP reply" packet, i.e., an ARP reply that specifies the
trailer encapsulation protocol type but otherwise has the
format of a normal ARP reply. If a host configured to use
trailers receives a trailer ARP reply message from a
remote machine, it can add that machine to the list of
machines that understand trailers, e.g., by marking the
corresponding entry in the ARP cache.
Hosts wishing to receive trailer encapsulations send
trailer ARP replies whenever they complete exchanges of
normal ARP messages for IP. Thus, a host that received an
ARP request for its IP protocol address would send a
trailer ARP reply in addition to the normal IP ARP reply;
a host that sent the IP ARP request would send a trailer
ARP reply when it received the corresponding IP ARP reply.
In this way, either the requesting or responding host in
an IP ARP exchange may request that it receive trailer
encapsulations.
This scheme, using extra trailer ARP reply packets rather
than sending an ARP request for the trailer protocol type,
was designed to avoid a continuous exchange of ARP packets
with a misbehaving host that, contrary to any
specification or common sense, responded to an ARP reply
for trailers with another ARP reply for IP. This problem
is avoided by sending a trailer ARP reply in response to
an IP ARP reply only when the IP ARP reply answers an
outstanding request; this is true when the hardware
address for the host is still unknown when the IP ARP
reply is received. A trailer ARP reply may always be sent
along with an IP ARP reply responding to an IP ARP
request.
2.3.2 Address Resolution Protocol -- ARP
2.3.2.1 ARP Cache Validation
An implementation of the Address Resolution Protocol (ARP)
[LINK:2] MUST provide a mechanism to flush out-of-date cache
entries. If this mechanism involves a timeout, it SHOULD be
possible to configure the timeout value.
A mechanism to prevent ARP flooding (repeatedly sending an
ARP Request for the same IP address, at a high rate) MUST be
included. The recommended maximum rate is 1 per second per
Internet Engineering Task Force [Page 22]
RFC1122 LINK LAYER October 1989
destination.
DISCUSSION:
The ARP specification [LINK:2] suggests but does not
require a timeout mechanism to invalidate cache entries
when hosts change their Ethernet addresses. The
prevalence of proxy ARP (see Section 2.4 of [INTRO:2])
has significantly increased the likelihood that cache
entries in hosts will become invalid, and therefore
some ARP-cache invalidation mechanism is now required
for hosts. Even in the absence of proxy ARP, a long-
period cache timeout is useful in order to
automatically correct any bad ARP data that might have
been cached.
IMPLEMENTATION:
Four mechanisms have been used, sometimes in
combination, to flush out-of-date cache entries.
(1) Timeout -- Periodically time out cache entries,
even if they are in use. Note that this timeout
should be restarted when the cache entry is
"refreshed" (by observing the source fields,
regardless of target address, of an ARP broadcast
from the system in question). For proxy ARP
situations, the timeout needs to be on the order
of a minute.
(2) Unicast Poll -- Actively poll the remote host by
periodically sending a point-to-point ARP Request
to it, and delete the entry if no ARP Reply is
received from N successive polls. Again, the
timeout should be on the order of a minute, and
typically N is 2.
(3) Link-Layer Advice -- If the link-layer driver
detects a delivery problem, flush the
corresponding ARP cache entry.
(4) Higher-layer Advice -- Provide a call from the
Internet layer to the link layer to indicate a
delivery problem. The effect of this call would
be to invalidate the corresponding cache entry.
This call would be analogous to the
"ADVISE_DELIVPROB()" call from the transport layer
to the Internet layer (see Section 3.4), and in
fact the ADVISE_DELIVPROB routine might in turn
call the link-layer advice routine to invalidate
Internet Engineering Task Force [Page 23]
RFC1122 LINK LAYER October 1989
the ARP cache entry.
Approaches (1) and (2) involve ARP cache timeouts on
the order of a minute or less. In the absence of proxy
ARP, a timeout this short could create noticeable
overhead traffic on a very large Ethernet. Therefore,
it may be necessary to configure a host to lengthen the
ARP cache timeout.
2.3.2.2 ARP Packet Queue
The link layer SHOULD save (rather than discard) at least
one (the latest) packet of each set of packets destined to
the same unresolved IP address, and transmit the saved
packet when the address has been resolved.
DISCUSSION:
Failure to follow this recommendation causes the first
packet of every exchange to be lost. Although higher-
layer protocols can generally cope with packet loss by
retransmission, packet loss does impact performance.
For example, loss of a TCP open request causes the
initial round-trip time estimate to be inflated. UDP-
based applications such as the Domain Name System are
more seriously affected.
2.3.3 Ethernet and IEEE 802 Encapsulation
The IP encapsulation for Ethernets is described in RFC-894
[LINK:3], while RFC-1042 [LINK:4] describes the IP
encapsulation for IEEE 802 networks. RFC-1042 elaborates and
replaces the discussion in Section 3.4 of [INTRO:2].
Every Internet host connected to a 10Mbps Ethernet cable:
o MUST be able to send and receive packets using RFC-894
encapsulation;
o SHOULD be able to receive RFC-1042 packets, intermixed
with RFC-894 packets; and
o MAY be able to send packets using RFC-1042 encapsulation.
An Internet host that implements sending both the RFC-894 and
the RFC-1042 encapsulations MUST provide a configuration switch
to select which is sent, and this switch MUST default to RFC-
894.
Internet Engineering Task Force [Page 24]
RFC1122 LINK LAYER October 1989
Note that the standard IP encapsulation in RFC-1042 does not
use the protocol id value (K1=6) that IEEE reserved for IP;
instead, it uses a value (K1=170) that implies an extension
(the "SNAP") which can be used to hold the Ether-Type field.
An Internet system MUST NOT send 802 packets using K1=6.
Address translation from Internet addresses to link-layer
addresses on Ethernet and IEEE 802 networks MUST be managed by
the Address Resolution Protocol (ARP).
The MTU for an Ethernet is 1500 and for 802.3 is 1492.
DISCUSSION:
The IEEE 802.3 specification provides for operation over a
10Mbps Ethernet cable, in which case Ethernet and IEEE
802.3 frames can be physically intermixed. A receiver can
distinguish Ethernet and 802.3 frames by the value of the
802.3 Length field; this two-octet field coincides in the
header with the Ether-Type field of an Ethernet frame. In
particular, the 802.3 Length field must be less than or
equal to 1500, while all valid Ether-Type values are
greater than 1500.
Another compatibility problem arises with link-layer
broadcasts. A broadcast sent with one framing will not be
seen by hosts that can receive only the other framing.
The provisions of this section were designed to provide
direct interoperation between 894-capable and 1042-capable
systems on the same cable, to the maximum extent possible.
It is intended to support the present situation where
894-only systems predominate, while providing an easy
transition to a possible future in which 1042-capable
systems become common.
Note that 894-only systems cannot interoperate directly
with 1042-only systems. If the two system types are set
up as two different logical networks on the same cable,
they can communicate only through an IP gateway.
Furthermore, it is not useful or even possible for a
dual-format host to discover automatically which format to
send, because of the problem of link-layer broadcasts.
2.4 LINK/INTERNET LAYER INTERFACE
The packet receive interface between the IP layer and the link
layer MUST include a flag to indicate whether the incoming packet
was addressed to a link-layer broadcast address.
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RFC1122 LINK LAYER October 1989
DISCUSSION
Although the IP layer does not generally know link layer
addresses (since every different network medium typically has
a different address format), the broadcast address on a
broadcast-capable medium is an important special case. See
Section 3.2.2, especially the DISCUSSION concerning broadcast
storms.
The packet send interface between the IP and link layers MUST
include the 5-bit TOS field (see Section 3.2.1.6).
The link layer MUST NOT report a Destination Unreachable error to
IP solely because there is no ARP cache entry for a destination.
2.5 LINK LAYER REQUIREMENTS SUMMARY
| | | | |S| |
| | | | |H| |F
| | | | |O|M|o
| | |S| |U|U|o
| | |H| |L|S|t
| |M|O| |D|T|n
| |U|U|M| | |o
| |S|L|A|N|N|t
| |T|D|Y|O|O|t
FEATURE |SECTION| | | |T|T|e
--------------------------------------------------|-------|-|-|-|-|-|--
| | | | | | |
Trailer encapsulation |2.3.1 | | |x| | |
Send Trailers by default without negotiation |2.3.1 | | | | |x|
ARP |2.3.2 | | | | | |
Flush out-of-date ARP cache entries |2.3.2.1|x| | | | |
Prevent ARP floods |2.3.2.1|x| | | | |
Cache timeout configurable |2.3.2.1| |x| | | |
Save at least one (latest) unresolved pkt |2.3.2.2| |x| | | |
Ethernet and IEEE 802 Encapsulation |2.3.3 | | | | | |
Host able to: |2.3.3 | | | | | |
Send & receive RFC-894 encapsulation |2.3.3 |x| | | | |
Receive RFC-1042 encapsulation |2.3.3 | |x| | | |
Send RFC-1042 encapsulation |2.3.3 | | |x| | |
Then config. sw. to select, RFC-894 dflt |2.3.3 |x| | | | |
Send K1=6 encapsulation |2.3.3 | | | | |x|
Use ARP on Ethernet and IEEE 802 nets |2.3.3 |x| | | | |
Link layer report b'casts to IP layer |2.4 |x| | | | |
IP layer pass TOS to link layer |2.4 |x| | | | |
No ARP cache entry treated as Dest. Unreach. |2.4 | | | | |x|
Internet Engineering Task Force [Page 26]
RFC1122 INTERNET LAYER October 1989
3. INTERNET LAYER PROTOCOLS
3.1 INTRODUCTION
The Robustness Principle: "Be liberal in what you accept, and
conservative in what you send" is particularly important in the
Internet layer, where one misbehaving host can deny Internet
service to many other hosts.
The protocol standards used in the Internet layer are:
o RFC-791 [IP:1] defines the IP protocol and gives an
introduction to the architecture of the Internet.
o RFC-792 [IP:2] defines ICMP, which provides routing,
diagnostic and error functionality for IP. Although ICMP
messages are encapsulated within IP datagrams, ICMP
processing is considered to be (and is typically implemented
as) part of the IP layer. See Section 3.2.2.
o RFC-950 [IP:3] defines the mandatory subnet extension to the
addressing architecture.
o RFC-1112 [IP:4] defines the Internet Group Management
Protocol IGMP, as part of a recommended extension to hosts
and to the host-gateway interface to support Internet-wide
multicasting at the IP level. See Section 3.2.3.
The target of an IP multicast may be an arbitrary group of
Internet hosts. IP multicasting is designed as a natural
extension of the link-layer multicasting facilities of some
networks, and it provides a standard means for local access
to such link-layer multicasting facilities.
Other important references are listed in Section 5 of this
document.
The Internet layer of host software MUST implement both IP and
ICMP. See Section 3.3.7 for the requirements on support of IGMP.
The host IP layer has two basic functions: (1) choose the "next
hop" gateway or host for outgoing IP datagrams and (2) reassemble
incoming IP datagrams. The IP layer may also (3) implement
intentional fragmentation of outgoing datagrams. Finally, the IP
layer must (4) provide diagnostic and error functionality. We
expect that IP layer functions may increase somewhat in the
future, as further Internet control and management facilities are
developed.
Internet Engineering Task Force [Page 27]
RFC1122 INTERNET LAYER October 1989
For normal datagrams, the processing is straightforward. For
incoming datagrams, the IP layer:
(1) verifies that the datagram is correctly formatted;
(2) verifies that it is destined to the local host;
(3) processes options;
(4) reassembles the datagram if necessary; and
(5) passes the encapsulated message to the appropriate
transport-layer protocol module.
For outgoing datagrams, the IP layer:
(1) sets any fields not set by the transport layer;
(2) selects the correct first hop on the connected network (a
process called "routing");
(3) fragments the datagram if necessary and if intentional
fragmentation is implemented (see Section 3.3.3); and
(4) passes the packet(s) to the appropriate link-layer driver.
A host is said to be multihomed if it has multiple IP addresses.
Multihoming introduces considerable confusion and complexity into
the protocol suite, and it is an area in which the Internet
architecture falls seriously short of solving all problems. There
are two distinct problem areas in multihoming:
(1) Local multihoming -- the host itself is multihomed; or
(2) Remote multihoming -- the local host needs to communicate
with a remote multihomed host.
At present, remote multihoming MUST be handled at the application
layer, as discussed in the companion RFC [INTRO:1]. A host MAY
support local multihoming, which is discussed in this document,
and in particular in Section 3.3.4.
Any host that forwards datagrams generated by another host is
acting as a gateway and MUST also meet the specifications laid out
in the gateway requirements RFC [INTRO:2]. An Internet host that
includes embedded gateway code MUST have a configuration switch to
disable the gateway function, and this switch MUST default to the
Internet Engineering Task Force [Page 28]
RFC1122 INTERNET LAYER October 1989
non-gateway mode. In this mode, a datagram arriving through one
interface will not be forwarded to another host or gateway (unless
it is source-routed), regardless of whether the host is single-
homed or multihomed. The host software MUST NOT automatically
move into gateway mode if the host has more than one interface, as
the operator of the machine may neither want to provide that
service nor be competent to do so.
In the following, the action specified in certain cases is to
"silently discard" a received datagram. This means that the
datagram will be discarded without further processing and that the
host will not send any ICMP error message (see Section 3.2.2) as a
result. However, for diagnosis of problems a host SHOULD provide
the capability of logging the error (see Section 1.2.3), including
the contents of the silently-discarded datagram, and SHOULD record
the event in a statistics counter.
DISCUSSION:
Silent discard of erroneous datagrams is generally intended
to prevent "broadcast storms".
3.2 PROTOCOL WALK-THROUGH
3.2.1 Internet Protocol -- IP
3.2.1.1 Version Number: RFC-791 Section 3.1
A datagram whose version number is not 4 MUST be silently
discarded.
3.2.1.2 Checksum: RFC-791 Section 3.1
A host MUST verify the IP header checksum on every received
datagram and silently discard every datagram that has a bad
checksum.
3.2.1.3 Addressing: RFC-791 Section 3.2
There are now five classes of IP addresses: Class A through
Class E. Class D addresses are used for IP multicasting
[IP:4], while Class E addresses are reserved for
experimental use.
A multicast (Class D) address is a 28-bit logical address
that stands for a group of hosts, and may be either
permanent or transient. Permanent multicast addresses are
allocated by the Internet Assigned Number Authority
[INTRO:6], while transient addresses may be allocated
Internet Engineering Task Force [Page 29]
RFC1122 INTERNET LAYER October 1989
dynamically to transient groups. Group membership is
determined dynamically using IGMP [IP:4].
We now summarize the important special cases for Class A, B,
and C IP addresses, using the following notation for an IP
address:
{ , }
or
{ , , }
and the notation "-1" for a field that contains all 1 bits.
This notation is not intended to imply that the 1-bits in an
address mask need be contiguous.
(a) { 0, 0 }
This host on this network. MUST NOT be sent, except as
a source address as part of an initialization procedure
by which the host learns its own IP address.
See also Section 3.3.6 for a non-standard use of {0,0}.
(b) { 0, }
Specified host on this network. It MUST NOT be sent,
except as a source address as part of an initialization
procedure by which the host learns its full IP address.
(c) { -1, -1 }
Limited broadcast. It MUST NOT be used as a source
address.
A datagram with this destination address will be
received by every host on the connected physical
network but will not be forwarded outside that network.
(d) { , -1 }
Directed broadcast to the specified network. It MUST
NOT be used as a source address.
(e) { , , -1 }
Directed broadcast to the specified subnet. It MUST
NOT be used as a source address.
Internet Engineering Task Force [Page 30]
RFC1122 INTERNET LAYER October 1989
(f) { , -1, -1 }
Directed broadcast to all subnets of the specified
subnetted network. It MUST NOT be used as a source
address.
(g) { 127, }
Internal host loopback address. Addresses of this form
MUST NOT appear outside a host.
The is administratively assigned so that
its value will be unique in the entire world.
IP addresses are not permitted to have the value 0 or -1 for
any of the , , or fields (except in the special cases listed above).
This implies that each of these fields will be at least two
bits long.
For further discussion of broadcast addresses, see Section
3.3.6.
A host MUST support the subnet extensions to IP [IP:3]. As
a result, there will be an address mask of the form:
{-1, -1, 0} associated with each of the host's local IP
addresses; see Sections 3.2.2.9 and 3.3.1.1.
When a host sends any datagram, the IP source address MUST
be one of its own IP addresses (but not a broadcast or
multicast address).
A host MUST silently discard an incoming datagram that is
not destined for the host. An incoming datagram is destined
for the host if the datagram's destination address field is:
(1) (one of) the host's IP address(es); or
(2) an IP broadcast address valid for the connected
network; or
(3) the address for a multicast group of which the host is
a member on the incoming physical interface.
For most purposes, a datagram addressed to a broadcast or
multicast destination is processed as if it had been
addressed to one of the host's IP addresses; we use the term
"specific-destination address" for the equivalent local IP
Internet Engineering Task Force [Page 31]
RFC1122 INTERNET LAYER October 1989
address of the host. The specific-destination address is
defined to be the destination address in the IP header
unless the header contains a broadcast or multicast address,
in which case the specific-destination is an IP address
assigned to the physical interface on which the datagram
arrived.
A host MUST silently discard an incoming datagram containing
an IP source address that is invalid by the rules of this
section. This validation could be done in either the IP
layer or by each protocol in the transport layer.
DISCUSSION:
A mis-addressed datagram might be caused by a link-
layer broadcast of a unicast datagram or by a gateway
or host that is confused or mis-configured.
An architectural goal for Internet hosts was to allow
IP addresses to be featureless 32-bit numbers, avoiding
algorithms that required a knowledge of the IP address
format. Otherwise, any future change in the format or
interpretation of IP addresses will require host
software changes. However, validation of broadcast and
multicast addresses violates this goal; a few other
violations are described elsewhere in this document.
Implementers should be aware that applications
depending upon the all-subnets directed broadcast
address (f) may be unusable on some networks. All-
subnets broadcast is not widely implemented in vendor
gateways at present, and even when it is implemented, a
particular network administration may disable it in the
gateway configuration.
3.2.1.4 Fragmentation and Reassembly: RFC-791 Section 3.2
The Internet model requires that every host support
reassembly. See Sections 3.3.2 and 3.3.3 for the
requirements on fragmentation and reassembly.
3.2.1.5 Identification: RFC-791 Section 3.2
When sending an identical copy of an earlier datagram, a
host MAY optionally retain the same Identification field in
the copy.
Internet Engineering Task Force [Page 32]
RFC1122 INTERNET LAYER October 1989
DISCUSSION:
Some Internet protocol experts have maintained that
when a host sends an identical copy of an earlier
datagram, the new copy should contain the same
Identification value as the original. There are two
suggested advantages: (1) if the datagrams are
fragmented and some of the fragments are lost, the
receiver may be able to reconstruct a complete datagram
from fragments of the original and the copies; (2) a
congested gateway might use the IP Identification field
(and Fragment Offset) to discard duplicate datagrams
from the queue.
However, the observed patterns of datagram loss in the
Internet do not favor the probability of retransmitted
fragments filling reassembly gaps, while other
mechanisms (e.g., TCP repacketizing upon
retransmission) tend to prevent retransmission of an
identical datagram [IP:9]. Therefore, we believe that
retransmitting the same Identification field is not
useful. Also, a connectionless transport protocol like
UDP would require the cooperation of the application
programs to retain the same Identification value in
identical datagrams.
3.2.1.6 Type-of-Service: RFC-791 Section 3.2
The "Type-of-Service" byte in the IP header is divided into
two sections: the Precedence field (high-order 3 bits), and
a field that is customarily called "Type-of-Service" or
"TOS" (low-order 5 bits). In this document, all references
to "TOS" or the "TOS field" refer to the low-order 5 bits
only.
The Precedence field is intended for Department of Defense
applications of the Internet protocols. The use of non-zero
values in this field is outside the scope of this document
and the IP standard specification. Vendors should consult
the Defense Communication Agency (DCA) for guidance on the
IP Precedence field and its implications for other protocol
layers. However, vendors should note that the use of
precedence will most likely require that its value be passed
between protocol layers in just the same way as the TOS
field is passed.
The IP layer MUST provide a means for the transport layer to
set the TOS field of every datagram that is sent; the
default is all zero bits. The IP layer SHOULD pass received
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TOS values up to the transport layer.
The particular link-layer mappings of TOS contained in RFC-
795 SHOULD NOT be implemented.
DISCUSSION:
While the TOS field has been little used in the past,
it is expected to play an increasing role in the near
future. The TOS field is expected to be used to
control two aspects of gateway operations: routing and
queueing algorithms. See Section 2 of [INTRO:1] for
the requirements on application programs to specify TOS
values.
The TOS field may also be mapped into link-layer
service selectors. This has been applied to provide
effective sharing of serial lines by different classes
of TCP traffic, for example. However, the mappings
suggested in RFC-795 for networks that were included in
the Internet as of 1981 are now obsolete.
3.2.1.7 Time-to-Live: RFC-791 Section 3.2
A host MUST NOT send a datagram with a Time-to-Live (TTL)
value of zero.
A host MUST NOT discard a datagram just because it was
received with TTL less than 2.
The IP layer MUST provide a means for the transport layer to
set the TTL field of every datagram that is sent. When a
fixed TTL value is used, it MUST be configurable. The
current suggested value will be published in the "Assigned
Numbers" RFC.
DISCUSSION:
The TTL field has two functions: limit the lifetime of
TCP segments (see RFC-793 [TCP:1], p. 28), and
terminate Internet routing loops. Although TTL is a
time in seconds, it also has some attributes of a hop-
count, since each gateway is required to reduce the TTL
field by at least one.
The intent is that TTL expiration will cause a datagram
to be discarded by a gateway but not by the destination
host; however, hosts that act as gateways by forwarding
datagrams must follow the gateway rules for TTL.
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A higher-layer protocol may want to set the TTL in
order to implement an "expanding scope" search for some
Internet resource. This is used by some diagnostic
tools, and is expected to be useful for locating the
"nearest" server of a given class using IP
multicasting, for example. A particular transport
protocol may also want to specify its own TTL bound on
maximum datagram lifetime.
A fixed value must be at least big enough for the
Internet "diameter," i.e., the longest possible path.
A reasonable value is about twice the diameter, to
allow for continued Internet growth.
3.2.1.8 Options: RFC-791 Section 3.2
There MUST be a means for the transport layer to specify IP
options to be included in transmitted IP datagrams (see
Section 3.4).
All IP options (except NOP or END-OF-LIST) received in
datagrams MUST be passed to the transport layer (or to ICMP
processing when the datagram is an ICMP message). The IP
and transport layer MUST each interpret those IP options
that they understand and silently ignore the others.
Later sections of this document discuss specific IP option
support required by each of ICMP, TCP, and UDP.
DISCUSSION:
Passing all received IP options to the transport layer
is a deliberate "violation of strict layering" that is
designed to ease the introduction of new transport-
relevant IP options in the future. Each layer must
pick out any options that are relevant to its own
processing and ignore the rest. For this purpose,
every IP option except NOP and END-OF-LIST will include
a specification of its own length.
This document does not define the order in which a
receiver must process multiple options in the same IP
header. Hosts sending multiple options must be aware
that this introduces an ambiguity in the meaning of
certain options when combined with a source-route
option.
IMPLEMENTATION:
The IP layer must not crash as the result of an option
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length that is outside the possible range. For
example, erroneous option lengths have been observed to
put some IP implementations into infinite loops.
Here are the requirements for specific IP options:
(a) Security Option
Some environments require the Security option in every
datagram; such a requirement is outside the scope of
this document and the IP standard specification. Note,
however, that the security options described in RFC-791
and RFC-1038 are obsolete. For DoD applications,
vendors should consult [IP:8] for guidance.
(b) Stream Identifier Option
This option is obsolete; it SHOULD NOT be sent, and it
MUST be silently ignored if received.
(c) Source Route Options
A host MUST support originating a source route and MUST
be able to act as the final destination of a source
route.
If host receives a datagram containing a completed
source route (i.e., the pointer points beyond the last
field), the datagram has reached its final destination;
the option as received (the recorded route) MUST be
passed up to the transport layer (or to ICMP message
processing). This recorded route will be reversed and
used to form a return source route for reply datagrams
(see discussion of IP Options in Section 4). When a
return source route is built, it MUST be correctly
formed even if the recorded route included the source
host (see case (B) in the discussion below).
An IP header containing more than one Source Route
option MUST NOT be sent; the effect on routing of
multiple Source Route options is implementation-
specific.
Section 3.3.5 presents the rules for a host acting as
an intermediate hop in a source route, i.e., forwarding
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a source-routed datagram.
DISCUSSION:
If a source-routed datagram is fragmented, each
fragment will contain a copy of the source route.
Since the processing of IP options (including a
source route) must precede reassembly, the
original datagram will not be reassembled until
the final destination is reached.
Suppose a source routed datagram is to be routed
from host S to host D via gateways G1, G2, ... Gn.
There was an ambiguity in the specification over
whether the source route option in a datagram sent
out by S should be (A) or (B):
(A): {>>G2, G3, ... Gn, D} >G2, G3, ... Gn, D} > represents the pointer). If (A) is
sent, the datagram received at D will contain the
option: {G1, G2, ... Gn >>}, with S and D as the
IP source and destination addresses. If (B) were
sent, the datagram received at D would again
contain S and D as the same IP source and
destination addresses, but the option would be:
{S, G1, ...Gn >>}; i.e., the originating host
would be the first hop in the route.
(d) Record Route Option
Implementation of originating and processing the Record
Route option is OPTIONAL.
(e) Timestamp Option
Implementation of originating and processing the
Timestamp option is OPTIONAL. If it is implemented,
the following rules apply:
o The originating host MUST record a timestamp in a
Timestamp option whose Internet address fields are
not pre-specified or whose first pre-specified
address is the host's interface address.
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o The destination host MUST (if possible) add the
current timestamp to a Timestamp option before
passing the option to the transport layer or to
ICMP for processing.
o A timestamp value MUST follow the rules given in
Section 3.2.2.8 for the ICMP Timestamp message.
3.2.2 Internet Control Message Protocol -- ICMP
ICMP messages are grouped into two classes.
*
ICMP error messages:
Destination Unreachable (see Section 3.2.2.1)
Redirect (see Section 3.2.2.2)
Source Quench (see Section 3.2.2.3)
Time Exceeded (see Section 3.2.2.4)
Parameter Problem (see Section 3.2.2.5)
*
ICMP query messages:
Echo (see Section 3.2.2.6)
Information (see Section 3.2.2.7)
Timestamp (see Section 3.2.2.8)
Address Mask (see Section 3.2.2.9)
If an ICMP message of unknown type is received, it MUST be
silently discarded.
Every ICMP error message includes the Internet header and at
least the first 8 data octets of the datagram that triggered
the error; more than 8 octets MAY be sent; this header and data
MUST be unchanged from the received datagram.
In those cases where the Internet layer is required to pass an
ICMP error message to the transport layer, the IP protocol
number MUST be extracted from the original header and used to
select the appropriate transport protocol entity to handle the
error.
An ICMP error message SHOULD be sent with normal (i.e., zero)
TOS bits.
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An ICMP error message MUST NOT be sent as the result of
receiving:
* an ICMP error message, or
* a datagram destined to an IP broadcast or IP multicast
address, or
* a datagram sent as a link-layer broadcast, or
* a non-initial fragment, or
* a datagram whose source address does not define a single
host -- e.g., a zero address, a loopback address, a
broadcast address, a multicast address, or a Class E
address.
NOTE: THESE RESTRICTIONS TAKE PRECEDENCE OVER ANY REQUIREMENT
ELSEWHERE IN THIS DOCUMENT FOR SENDING ICMP ERROR MESSAGES.
DISCUSSION:
These rules will prevent the "broadcast storms" that have
resulted from hosts returning ICMP error messages in
response to broadcast datagrams. For example, a broadcast
UDP segment to a non-existent port could trigger a flood
of ICMP Destination Unreachable datagrams from all
machines that do not have a client for that destination
port. On a large Ethernet, the resulting collisions can
render the network useless for a second or more.
Every datagram that is broadcast on the connected network
should have a valid IP broadcast address as its IP
destination (see Section 3.3.6). However, some hosts
violate this rule. To be certain to detect broadcast
datagrams, therefore, hosts are required to check for a
link-layer broadcast as well as an IP-layer broadcast
address.
IMPLEMENTATION:
This requires that the link layer inform the IP layer when
a link-layer broadcast datagram has been received; see
Section 2.4.
3.2.2.1 Destination Unreachable: RFC-792
The following additional codes are hereby defined:
6 = destination network unknown
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7 = destination host unknown
8 = source host isolated
9 = communication with destination network
administratively prohibited
10 = communication with destination host
administratively prohibited
11 = network unreachable for type of service
12 = host unreachable for type of service
A host SHOULD generate Destination Unreachable messages with
code:
2 (Protocol Unreachable), when the designated transport
protocol is not supported; or
3 (Port Unreachable), when the designated transport
protocol (e.g., UDP) is unable to demultiplex the
datagram but has no protocol mechanism to inform the
sender.
A Destination Unreachable message that is received MUST be
reported to the transport layer. The transport layer SHOULD
use the information appropriately; for example, see Sections
4.1.3.3, 4.2.3.9, and 4.2.4 below. A transport protocol
that has its own mechanism for notifying the sender that a
port is unreachable (e.g., TCP, which sends RST segments)
MUST nevertheless accept an ICMP Port Unreachable for the
same purpose.
A Destination Unreachable message that is received with code
0 (Net), 1 (Host), or 5 (Bad Source Route) may result from a
routing transient and MUST therefore be interpreted as only
a hint, not proof, that the specified destination is
unreachable [IP:11]. For example, it MUST NOT be used as
proof of a dead gateway (see Section 3.3.1).
3.2.2.2 Redirect: RFC-792
A host SHOULD NOT send an ICMP Redirect message; Redirects
are to be sent only by gateways.
A host receiving a Redirect message MUST update its routing
information accordingly. Every host MUST be prepared to
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accept both Host and Network Redirects and to process them
as described in Section 3.3.1.2 below.
A Redirect message SHOULD be silently discarded if the new
gateway address it specifies is not on the same connected
(sub-) net through which the Redirect arrived [INTRO:2,
Appendix A], or if the source of the Redirect is not the
current first-hop gateway for the specified destination (see
Section 3.3.1).
3.2.2.3 Source Quench: RFC-792
A host MAY send a Source Quench message if it is
approaching, or has reached, the point at which it is forced
to discard incoming datagrams due to a shortage of
reassembly buffers or other resources. See Section 2.2.3 of
[INTRO:2] for suggestions on when to send Source Quench.
If a Source Quench message is received, the IP layer MUST
report it to the transport layer (or ICMP processing). In
general, the transport or application layer SHOULD implement
a mechanism to respond to Source Quench for any protocol
that can send a sequence of datagrams to the same
destination and which can reasonably be expected to maintain
enough state information to make this feasible. See Section
4 for the handling of Source Quench by TCP and UDP.
DISCUSSION:
A Source Quench may be generated by the target host or
by some gateway in the path of a datagram. The host
receiving a Source Quench should throttle itself back
for a period of time, then gradually increase the
transmission rate again. The mechanism to respond to
Source Quench may be in the transport layer (for
connection-oriented protocols like TCP) or in the
application layer (for protocols that are built on top
of UDP).
A mechanism has been proposed [IP:14] to make the IP
layer respond directly to Source Quench by controlling
the rate at which datagrams are sent, however, this
proposal is currently experimental and not currently
recommended.
3.2.2.4 Time Exceeded: RFC-792
An incoming Time Exceeded message MUST be passed to the
transport layer.
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DISCUSSION:
A gateway will send a Time Exceeded Code 0 (In Transit)
message when it discards a datagram due to an expired
TTL field. This indicates either a gateway routing
loop or too small an initial TTL value.
A host may receive a Time Exceeded Code 1 (Reassembly
Timeout) message from a destination host that has timed
out and discarded an incomplete datagram; see Section
3.3.2 below. In the future, receipt of this message
might be part of some "MTU discovery" procedure, to
discover the maximum datagram size that can be sent on
the path without fragmentation.
3.2.2.5 Parameter Problem: RFC-792
A host SHOULD generate Parameter Problem messages. An
incoming Parameter Problem message MUST be passed to the
transport layer, and it MAY be reported to the user.
DISCUSSION:
The ICMP Parameter Problem message is sent to the
source host for any problem not specifically covered by
another ICMP message. Receipt of a Parameter Problem
message generally indicates some local or remote
implementation error.
A new variant on the Parameter Problem message is hereby
defined:
Code 1 = required option is missing.
DISCUSSION:
This variant is currently in use in the military
community for a missing security option.
3.2.2.6 Echo Request/Reply: RFC-792
Every host MUST implement an ICMP Echo server function that
receives Echo Requests and sends corresponding Echo Replies.
A host SHOULD also implement an application-layer interface
for sending an Echo Request and receiving an Echo Reply, for
diagnostic purposes.
An ICMP Echo Request destined to an IP broadcast or IP
multicast address MAY be silently discarded.
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DISCUSSION:
This neutral provision results from a passionate debate
between those who feel that ICMP Echo to a broadcast
address provides a valuable diagnostic capability and
those who feel that misuse of this feature can too
easily create packet storms.
The IP source address in an ICMP Echo Reply MUST be the same
as the specific-destination address (defined in Section
3.2.1.3) of the corresponding ICMP Echo Request message.
Data received in an ICMP Echo Request MUST be entirely
included in the resulting Echo Reply. However, if sending
the Echo Reply requires intentional fragmentation that is
not implemented, the datagram MUST be truncated to maximum
transmission size (see Section 3.3.3) and sent.
Echo Reply messages MUST be passed to the ICMP user
interface, unless the corresponding Echo Request originated
in the IP layer.
If a Record Route and/or Time Stamp option is received in an
ICMP Echo Request, this option (these options) SHOULD be
updated to include the current host and included in the IP
header of the Echo Reply message, without "truncation".
Thus, the recorded route will be for the entire round trip.
If a Source Route option is received in an ICMP Echo
Request, the return route MUST be reversed and used as a
Source Route option for the Echo Reply message.
3.2.2.7 Information Request/Reply: RFC-792
A host SHOULD NOT implement these messages.
DISCUSSION:
The Information Request/Reply pair was intended to
support self-configuring systems such as diskless
workstations, to allow them to discover their IP
network numbers at boot time. However, the RARP and
BOOTP protocols provide better mechanisms for a host to
discover its own IP address.
3.2.2.8 Timestamp and Timestamp Reply: RFC-792
A host MAY implement Timestamp and Timestamp Reply. If they
are implemented, the following rules MUST be followed.
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o The ICMP Timestamp server function returns a Timestamp
Reply to every Timestamp message that is received. If
this function is implemented, it SHOULD be designed for
minimum variability in delay (e.g., implemented in the
kernel to avoid delay in scheduling a user process).
The following cases for Timestamp are to be handled
according to the corresponding rules for ICMP Echo:
o An ICMP Timestamp Request message to an IP broadcast or
IP multicast address MAY be silently discarded.
o The IP source address in an ICMP Timestamp Reply MUST
be the same as the specific-destination address of the
corresponding Timestamp Request message.
o If a Source-route option is received in an ICMP Echo
Request, the return route MUST be reversed and used as
a Source Route option for the Timestamp Reply message.
o If a Record Route and/or Timestamp option is received
in a Timestamp Request, this (these) option(s) SHOULD
be updated to include the current host and included in
the IP header of the Timestamp Reply message.
o Incoming Timestamp Reply messages MUST be passed up to
the ICMP user interface.
The preferred form for a timestamp value (the "standard
value") is in units of milliseconds since midnight Universal
Time. However, it may be difficult to provide this value
with millisecond resolution. For example, many systems use
clocks that update only at line frequency, 50 or 60 times
per second. Therefore, some latitude is allowed in a
"standard value":
(a) A "standard value" MUST be updated at least 15 times
per second (i.e., at most the six low-order bits of the
value may be undefined).
(b) The accuracy of a "standard value" MUST approximate
that of operator-set CPU clocks, i.e., correct within a
few minutes.
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3.2.2.9 Address Mask Request/Reply: RFC-950
A host MUST support the first, and MAY implement all three,
of the following methods for determining the address mask(s)
corresponding to its IP address(es):
(1) static configuration information;
(2) obtaining the address mask(s) dynamically as a side-
effect of the system initialization process (see
[INTRO:1]); and
(3) sending ICMP Address Mask Request(s) and receiving ICMP
Address Mask Reply(s).
The choice of method to be used in a particular host MUST be
configurable.
When method (3), the use of Address Mask messages, is
enabled, then:
(a) When it initializes, the host MUST broadcast an Address
Mask Request message on the connected network
corresponding to the IP address. It MUST retransmit
this message a small number of times if it does not
receive an immediate Address Mask Reply.
(b) Until it has received an Address Mask Reply, the host
SHOULD assume a mask appropriate for the address class
of the IP address, i.e., assume that the connected
network is not subnetted.
(c) The first Address Mask Reply message received MUST be
used to set the address mask corresponding to the
particular local IP address. This is true even if the
first Address Mask Reply message is "unsolicited", in
which case it will have been broadcast and may arrive
after the host has ceased to retransmit Address Mask
Requests. Once the mask has been set by an Address
Mask Reply, later Address Mask Reply messages MUST be
(silently) ignored.
Conversely, if Address Mask messages are disabled, then no
ICMP Address Mask Requests will be sent, and any ICMP
Address Mask Replies received for that local IP address MUST
be (silently) ignored.
A host SHOULD make some reasonableness check on any address
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mask it installs; see IMPLEMENTATION section below.
A system MUST NOT send an Address Mask Reply unless it is an
authoritative agent for address masks. An authoritative
agent may be a host or a gateway, but it MUST be explicitly
configured as a address mask agent. Receiving an address
mask via an Address Mask Reply does not give the receiver
authority and MUST NOT be used as the basis for issuing
Address Mask Replies.
With a statically configured address mask, there SHOULD be
an additional configuration flag that determines whether the
host is to act as an authoritative agent for this mask,
i.e., whether it will answer Address Mask Request messages
using this mask.
If it is configured as an agent, the host MUST broadcast an
Address Mask Reply for the mask on the appropriate interface
when it initializes.
See "System Initialization" in [INTRO:1] for more
information about the use of Address Mask Request/Reply
messages.
DISCUSSION
Hosts that casually send Address Mask Replies with
invalid address masks have often been a serious
nuisance. To prevent this, Address Mask Replies ought
to be sent only by authoritative agents that have been
selected by explicit administrative action.
When an authoritative agent receives an Address Mask
Request message, it will send a unicast Address Mask
Reply to the source IP address. If the network part of
this address is zero (see (a) and (b) in 3.2.1.3), the
Reply will be broadcast.
Getting no reply to its Address Mask Request messages,
a host will assume there is no agent and use an
unsubnetted mask, but the agent may be only temporarily
unreachable. An agent will broadcast an unsolicited
Address Mask Reply whenever it initializes, in order to
update the masks of all hosts that have initialized in
the meantime.
IMPLEMENTATION:
The following reasonableness check on an address mask
is suggested: the mask is not all 1 bits, and it is
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either zero or else the 8 highest-order bits are on.
3.2.3 Internet Group Management Protocol IGMP
IGMP [IP:4] is a protocol used between hosts and gateways on a
single network to establish hosts' membership in particular
multicast groups. The gateways use this information, in
conjunction with a multicast routing protocol, to support IP
multicasting across the Internet.
At this time, implementation of IGMP is OPTIONAL; see Section
3.3.7 for more information. Without IGMP, a host can still
participate in multicasting local to its connected networks.
3.3 SPECIFIC ISSUES
3.3.1 Routing Outbound Datagrams
The IP layer chooses the correct next hop for each datagram it
sends. If the destination is on a connected network, the
datagram is sent directly to the destination host; otherwise,
it has to be routed to a gateway on a connected network.
3.3.1.1 Local/Remote Decision
To decide if the destination is on a connected network, the
following algorithm MUST be used [see IP:3]:
(a) The address mask (particular to a local IP address for
a multihomed host) is a 32-bit mask that selects the
network number and subnet number fields of the
corresponding IP address.
(b) If the IP destination address bits extracted by the
address mask match the IP source address bits extracted
by the same mask, then the destination is on the
corresponding connected network, and the datagram is to
be transmitted directly to the destination host.
(c) If not, then the destination is accessible only through
a gateway. Selection of a gateway is described below
(3.3.1.2).
A special-case destination address is handled as follows:
* For a limited broadcast or a multicast address, simply
pass the datagram to the link layer for the appropriate
interface.
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* For a (network or subnet) directed broadcast, the
datagram can use the standard routing algorithms.
The host IP layer MUST operate correctly in a minimal
network environment, and in particular, when there are no
gateways. For example, if the IP layer of a host insists on
finding at least one gateway to initialize, the host will be
unable to operate on a single isolated broadcast net.
3.3.1.2 Gateway Selection
To efficiently route a series of datagrams to the same
destination, the source host MUST keep a "route cache" of
mappings to next-hop gateways. A host uses the following
basic algorithm on this cache to route a datagram; this
algorithm is designed to put the primary routing burden on
the gateways [IP:11].
(a) If the route cache contains no information for a
particular destination, the host chooses a "default"
gateway and sends the datagram to it. It also builds a
corresponding Route Cache entry.
(b) If that gateway is not the best next hop to the
destination, the gateway will forward the datagram to
the best next-hop gateway and return an ICMP Redirect
message to the source host.
(c) When it receives a Redirect, the host updates the
next-hop gateway in the appropriate route cache entry,
so later datagrams to the same destination will go
directly to the best gateway.
Since the subnet mask appropriate to the destination address
is generally not known, a Network Redirect message SHOULD be
treated identically to a Host Redirect message; i.e., the
cache entry for the destination host (only) would be updated
(or created, if an entry for that host did not exist) for
the new gateway.
DISCUSSION:
This recommendation is to protect against gateways that
erroneously send Network Redirects for a subnetted
network, in violation of the gateway requirements
[INTRO:2].
When there is no route cache entry for the destination host
address (and the destination is not on the connected
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network), the IP layer MUST pick a gateway from its list of
"default" gateways. The IP layer MUST support multiple
default gateways.
As an extra feature, a host IP layer MAY implement a table
of "static routes". Each such static route MAY include a
flag specifying whether it may be overridden by ICMP
Redirects.
DISCUSSION:
A host generally needs to know at least one default
gateway to get started. This information can be
obtained from a configuration file or else from the
host startup sequence, e.g., the BOOTP protocol (see
[INTRO:1]).
It has been suggested that a host can augment its list
of default gateways by recording any new gateways it
learns about. For example, it can record every gateway
to which it is ever redirected. Such a feature, while
possibly useful in some circumstances, may cause
problems in other cases (e.g., gateways are not all
equal), and it is not recommended.
A static route is typically a particular preset mapping
from destination host or network into a particular
next-hop gateway; it might also depend on the Type-of-
Service (see next section). Static routes would be set
up by system administrators to override the normal
automatic routing mechanism, to handle exceptional
situations. However, any static routing information is
a potential source of failure as configurations change
or equipment fails.
3.3.1.3 Route Cache
Each route cache entry needs to include the following
fields:
(1) Local IP address (for a multihomed host)
(2) Destination IP address
(3) Type(s)-of-Service
(4) Next-hop gateway IP address
Field (2) MAY be the full IP address of the destination
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host, or only the destination network number. Field (3),
the TOS, SHOULD be included.
See Section 3.3.4.2 for a discussion of the implications of
multihoming for the lookup procedure in this cache.
DISCUSSION:
Including the Type-of-Service field in the route cache
and considering it in the host route algorithm will
provide the necessary mechanism for the future when
Type-of-Service routing is commonly used in the
Internet. See Section 3.2.1.6.
Each route cache entry defines the endpoints of an
Internet path. Although the connecting path may change
dynamically in an arbitrary way, the transmission
characteristics of the path tend to remain
approximately constant over a time period longer than a
single typical host-host transport connection.
Therefore, a route cache entry is a natural place to
cache data on the properties of the path. Examples of
such properties might be the maximum unfragmented
datagram size (see Section 3.3.3), or the average
round-trip delay measured by a transport protocol.
This data will generally be both gathered and used by a
higher layer protocol, e.g., by TCP, or by an
application using UDP. Experiments are currently in
progress on caching path properties in this manner.
There is no consensus on whether the route cache should
be keyed on destination host addresses alone, or allow
both host and network addresses. Those who favor the
use of only host addresses argue that:
(1) As required in Section 3.3.1.2, Redirect messages
will generally result in entries keyed on
destination host addresses; the simplest and most
general scheme would be to use host addresses
always.
(2) The IP layer may not always know the address mask
for a network address in a complex subnetted
environment.
(3) The use of only host addresses allows the
destination address to be used as a pure 32-bit
number, which may allow the Internet architecture
to be more easily extended in the future without
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any change to the hosts.
The opposing view is that allowing a mixture of
destination hosts and networks in the route cache:
(1) Saves memory space.
(2) Leads to a simpler data structure, easily
combining the cache with the tables of default and
static routes (see below).
(3) Provides a more useful place to cache path
properties, as discussed earlier.
IMPLEMENTATION:
The cache needs to be large enough to include entries
for the maximum number of destination hosts that may be
in use at one time.
A route cache entry may also include control
information used to choose an entry for replacement.
This might take the form of a "recently used" bit, a
use count, or a last-used timestamp, for example. It
is recommended that it include the time of last
modification of the entry, for diagnostic purposes.
An implementation may wish to reduce the overhead of
scanning the route cache for every datagram to be
transmitted. This may be accomplished with a hash
table to speed the lookup, or by giving a connection-
oriented transport protocol a "hint" or temporary
handle on the appropriate cache entry, to be passed to
the IP layer with each subsequent datagram.
Although we have described the route cache, the lists
of default gateways, and a table of static routes as
conceptually distinct, in practice they may be combined
into a single "routing table" data structure.
3.3.1.4 Dead Gateway Detection
The IP layer MUST be able to detect the failure of a "next-
hop" gateway that is listed in its route cache and to choose
an alternate gateway (see Section 3.3.1.5).
Dead gateway detection is covered in some detail in RFC-816
[IP:11]. Experience to date has not produced a complete
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algorithm which is totally satisfactory, though it has
identified several forbidden paths and promising techniques.
* A particular gateway SHOULD NOT be used indefinitely in
the absence of positive indications that it is
functioning.
* Active probes such as "pinging" (i.e., using an ICMP
Echo Request/Reply exchange) are expensive and scale
poorly. In particular, hosts MUST NOT actively check
the status of a first-hop gateway by simply pinging the
gateway continuously.
* Even when it is the only effective way to verify a
gateway's status, pinging MUST be used only when
traffic is being sent to the gateway and when there is
no other positive indication to suggest that the
gateway is functioning.
* To avoid pinging, the layers above and/or below the
Internet layer SHOULD be able to give "advice" on the
status of route cache entries when either positive
(gateway OK) or negative (gateway dead) information is
available.
DISCUSSION:
If an implementation does not include an adequate
mechanism for detecting a dead gateway and re-routing,
a gateway failure may cause datagrams to apparently
vanish into a "black hole". This failure can be
extremely confusing for users and difficult for network
personnel to debug.
The dead-gateway detection mechanism must not cause
unacceptable load on the host, on connected networks,
or on first-hop gateway(s). The exact constraints on
the timeliness of dead gateway detection and on
acceptable load may vary somewhat depending on the
nature of the host's mission, but a host generally
needs to detect a failed first-hop gateway quickly
enough that transport-layer connections will not break
before an alternate gateway can be selected.
Passing advice from other layers of the protocol stack
complicates the interfaces between the layers, but it
is the preferred approach to dead gateway detection.
Advice can come from almost any part of the IP/TCP
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architecture, but it is expected to come primarily from
the transport and link layers. Here are some possible
sources for gateway advice:
o TCP or any connection-oriented transport protocol
should be able to give negative advice, e.g.,
triggered by excessive retransmissions.
o TCP may give positive advice when (new) data is
acknowledged. Even though the route may be
asymmetric, an ACK for new data proves that the
acknowleged data must have been transmitted
successfully.
o An ICMP Redirect message from a particular gateway
should be used as positive advice about that
gateway.
o Link-layer information that reliably detects and
reports host failures (e.g., ARPANET Destination
Dead messages) should be used as negative advice.
o Failure to ARP or to re-validate ARP mappings may
be used as negative advice for the corresponding
IP address.
o Packets arriving from a particular link-layer
address are evidence that the system at this
address is alive. However, turning this
information into advice about gateways requires
mapping the link-layer address into an IP address,
and then checking that IP address against the
gateways pointed to by the route cache. This is
probably prohibitively inefficient.
Note that positive advice that is given for every
datagram received may cause unacceptable overhead in
the implementation.
While advice might be passed using required arguments
in all interfaces to the IP layer, some transport and
application layer protocols cannot deduce the correct
advice. These interfaces must therefore allow a
neutral value for advice, since either always-positive
or always-negative advice leads to incorrect behavior.
There is another technique for dead gateway detection
that has been commonly used but is not recommended.
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This technique depends upon the host passively
receiving ("wiretapping") the Interior Gateway Protocol
(IGP) datagrams that the gateways are broadcasting to
each other. This approach has the drawback that a host
needs to recognize all the interior gateway protocols
that gateways may use (see [INTRO:2]). In addition, it
only works on a broadcast network.
At present, pinging (i.e., using ICMP Echo messages) is
the mechanism for gateway probing when absolutely
required. A successful ping guarantees that the
addressed interface and its associated machine are up,
but it does not guarantee that the machine is a gateway
as opposed to a host. The normal inference is that if
a Redirect or other evidence indicates that a machine
was a gateway, successful pings will indicate that the
machine is still up and hence still a gateway.
However, since a host silently discards packets that a
gateway would forward or redirect, this assumption
could sometimes fail. To avoid this problem, a new
ICMP message under development will ask "are you a
gateway?"
IMPLEMENTATION:
The following specific algorithm has been suggested:
o Associate a "reroute timer" with each gateway
pointed to by the route cache. Initialize the
timer to a value Tr, which must be small enough to
allow detection of a dead gateway before transport
connections time out.
o Positive advice would reset the reroute timer to
Tr. Negative advice would reduce or zero the
reroute timer.
o Whenever the IP layer used a particular gateway to
route a datagram, it would check the corresponding
reroute timer. If the timer had expired (reached
zero), the IP layer would send a ping to the
gateway, followed immediately by the datagram.
o The ping (ICMP Echo) would be sent again if
necessary, up to N times. If no ping reply was
received in N tries, the gateway would be assumed
to have failed, and a new first-hop gateway would
be chosen for all cache entries pointing to the
failed gateway.
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Note that the size of Tr is inversely related to the
amount of advice available. Tr should be large enough
to insure that:
* Any pinging will be at a low level (e.g., <10%) of
all packets sent to a gateway from the host, AND
* pinging is infrequent (e.g., every 3 minutes)
Since the recommended algorithm is concerned with the
gateways pointed to by route cache entries, rather than
the cache entries themselves, a two level data
structure (perhaps coordinated with ARP or similar
caches) may be desirable for implementing a route
cache.
3.3.1.5 New Gateway Selection
If the failed gateway is not the current default, the IP
layer can immediately switch to a default gateway. If it is
the current default that failed, the IP layer MUST select a
different default gateway (assuming more than one default is
known) for the failed route and for establishing new routes.
DISCUSSION:
When a gateway does fail, the other gateways on the
connected network will learn of the failure through
some inter-gateway routing protocol. However, this
will not happen instantaneously, since gateway routing
protocols typically have a settling time of 30-60
seconds. If the host switches to an alternative
gateway before the gateways have agreed on the failure,
the new target gateway will probably forward the
datagram to the failed gateway and send a Redirect back
to the host pointing to the failed gateway (!). The
result is likely to be a rapid oscillation in the
contents of the host's route cache during the gateway
settling period. It has been proposed that the dead-
gateway logic should include some hysteresis mechanism
to prevent such oscillations. However, experience has
not shown any harm from such oscillations, since
service cannot be restored to the host until the
gateways' routing information does settle down.
IMPLEMENTATION:
One implementation technique for choosing a new default
gateway is to simply round-robin among the default
gateways in the host's list. Another is to rank the
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gateways in priority order, and when the current
default gateway is not the highest priority one, to
"ping" the higher-priority gateways slowly to detect
when they return to service. This pinging can be at a
very low rate, e.g., 0.005 per second.
3.3.1.6 Initialization
The following information MUST be configurable:
(1) IP address(es).
(2) Address mask(s).
(3) A list of default gateways, with a preference level.
A manual method of entering this configuration data MUST be
provided. In addition, a variety of methods can be used to
determine this information dynamically; see the section on
"Host Initialization" in [INTRO:1].
DISCUSSION:
Some host implementations use "wiretapping" of gateway
protocols on a broadcast network to learn what gateways
exist. A standard method for default gateway discovery
is under development.
3.3.2 Reassembly
The IP layer MUST implement reassembly of IP datagrams.
We designate the largest datagram size that can be reassembled
by EMTU_R ("Effective MTU to receive"); this is sometimes
called the "reassembly buffer size". EMTU_R MUST be greater
than or equal to 576, SHOULD be either configurable or
indefinite, and SHOULD be greater than or equal to the MTU of
the connected network(s).
DISCUSSION:
A fixed EMTU_R limit should not be built into the code
because some application layer protocols require EMTU_R
values larger than 576.
IMPLEMENTATION:
An implementation may use a contiguous reassembly buffer
for each datagram, or it may use a more complex data
structure that places no definite limit on the reassembled
datagram size; in the latter case, EMTU_R is said to be
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"indefinite".
Logically, reassembly is performed by simply copying each
fragment into the packet buffer at the proper offset.
Note that fragments may overlap if successive
retransmissions use different packetizing but the same
reassembly Id.
The tricky part of reassembly is the bookkeeping to
determine when all bytes of the datagram have been
reassembled. We recommend Clark's algorithm [IP:10] that
requires no additional data space for the bookkeeping.
However, note that, contrary to [IP:10], the first
fragment header needs to be saved for inclusion in a
possible ICMP Time Exceeded (Reassembly Timeout) message.
There MUST be a mechanism by which the transport layer can
learn MMS_R, the maximum message size that can be received and
reassembled in an IP datagram (see GET_MAXSIZES calls in
Section 3.4). If EMTU_R is not indefinite, then the value of
MMS_R is given by:
MMS_R = EMTU_R - 20
since 20 is the minimum size of an IP header.
There MUST be a reassembly timeout. The reassembly timeout
value SHOULD be a fixed value, not set from the remaining TTL.
It is recommended that the value lie between 60 seconds and 120
seconds. If this timeout expires, the partially-reassembled
datagram MUST be discarded and an ICMP Time Exceeded message
sent to the source host (if fragment zero has been received).
DISCUSSION:
The IP specification says that the reassembly timeout
should be the remaining TTL from the IP header, but this
does not work well because gateways generally treat TTL as
a simple hop count rather than an elapsed time. If the
reassembly timeout is too small, datagrams will be
discarded unnecessarily, and communication may fail. The
timeout needs to be at least as large as the typical
maximum delay across the Internet. A realistic minimum
reassembly timeout would be 60 seconds.
It has been suggested that a cache might be kept of
round-trip times measured by transport protocols for
various destinations, and that these values might be used
to dynamically determine a reasonable reassembly timeout
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value. Further investigation of this approach is
required.
If the reassembly timeout is set too high, buffer
resources in the receiving host will be tied up too long,
and the MSL (Maximum Segment Lifetime) [TCP:1] will be
larger than necessary. The MSL controls the maximum rate
at which fragmented datagrams can be sent using distinct
values of the 16-bit Ident field; a larger MSL lowers the
maximum rate. The TCP specification [TCP:1] arbitrarily
assumes a value of 2 minutes for MSL. This sets an upper
limit on a reasonable reassembly timeout value.
3.3.3 Fragmentation
Optionally, the IP layer MAY implement a mechanism to fragment
outgoing datagrams intentionally.
We designate by EMTU_S ("Effective MTU for sending") the
maximum IP datagram size that may be sent, for a particular
combination of IP source and destination addresses and perhaps
TOS.
A host MUST implement a mechanism to allow the transport layer
to learn MMS_S, the maximum transport-layer message size that
may be sent for a given {source, destination, TOS} triplet (see
GET_MAXSIZES call in Section 3.4). If no local fragmentation
is performed, the value of MMS_S will be:
MMS_S = EMTU_S -
and EMTU_S must be less than or equal to the MTU of the network
interface corresponding to the source address of the datagram.
Note that in this equation will be 20, unless
the IP reserves space to insert IP options for its own purposes
in addition to any options inserted by the transport layer.
A host that does not implement local fragmentation MUST ensure
that the transport layer (for TCP) or the application layer
(for UDP) obtains MMS_S from the IP layer and does not send a
datagram exceeding MMS_S in size.
It is generally desirable to avoid local fragmentation and to
choose EMTU_S low enough to avoid fragmentation in any gateway
along the path. In the absence of actual knowledge of the
minimum MTU along the path, the IP layer SHOULD use
EMTU_S <= 576 whenever the destination address is not on a
connected network, and otherwise use the connected network's
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MTU.
The MTU of each physical interface MUST be configurable.
A host IP layer implementation MAY have a configuration flag
"All-Subnets-MTU", indicating that the MTU of the connected
network is to be used for destinations on different subnets
within the same network, but not for other networks. Thus,
this flag causes the network class mask, rather than the subnet
address mask, to be used to choose an EMTU_S. For a multihomed
host, an "All-Subnets-MTU" flag is needed for each network
interface.
DISCUSSION:
Picking the correct datagram size to use when sending data
is a complex topic [IP:9].
(a) In general, no host is required to accept an IP
datagram larger than 576 bytes (including header and
data), so a host must not send a larger datagram
without explicit knowledge or prior arrangement with
the destination host. Thus, MMS_S is only an upper
bound on the datagram size that a transport protocol
may send; even when MMS_S exceeds 556, the transport
layer must limit its messages to 556 bytes in the
absence of other knowledge about the destination
host.
(b) Some transport protocols (e.g., TCP) provide a way to
explicitly inform the sender about the largest
datagram the other end can receive and reassemble
[IP:7]. There is no corresponding mechanism in the
IP layer.
A transport protocol that assumes an EMTU_R larger
than 576 (see Section 3.3.2), can send a datagram of
this larger size to another host that implements the
same protocol.
(c) Hosts should ideally limit their EMTU_S for a given
destination to the minimum MTU of all the networks
along the path, to avoid any fragmentation. IP
fragmentation, while formally correct, can create a
serious transport protocol performance problem,
because loss of a single fragment means all the
fragments in the segment must be retransmitted
[IP:9].
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Since nearly all networks in the Internet currently
support an MTU of 576 or greater, we strongly recommend
the use of 576 for datagrams sent to non-local networks.
It has been suggested that a host could determine the MTU
over a given path by sending a zero-offset datagram
fragment and waiting for the receiver to time out the
reassembly (which cannot complete!) and return an ICMP
Time Exceeded message. This message would include the
largest remaining fragment header in its body. More
direct mechanisms are being experimented with, but have
not yet been adopted (see e.g., RFC-1063).
3.3.4 Local Multihoming
3.3.4.1 Introduction
A multihomed host has multiple IP addresses, which we may
think of as "logical interfaces". These logical interfaces
may be associated with one or more physical interfaces, and
these physical interfaces may be connected to the same or
different networks.
Here are some important cases of multihoming:
(a) Multiple Logical Networks
The Internet architects envisioned that each physical
network would have a single unique IP network (or
subnet) number. However, LAN administrators have
sometimes found it useful to violate this assumption,
operating a LAN with multiple logical networks per
physical connected network.
If a host connected to such a physical network is
configured to handle traffic for each of N different
logical networks, then the host will have N logical
interfaces. These could share a single physical
interface, or might use N physical interfaces to the
same network.
(b) Multiple Logical Hosts
When a host has multiple IP addresses that all have the
same part (and the same part, if any), the logical interfaces are known
as "logical hosts". These logical interfaces might
share a single physical interface or might use separate
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physical interfaces to the same physical network.
(c) Simple Multihoming
In this case, each logical interface is mapped into a
separate physical interface and each physical interface
is connected to a different physical network. The term
"multihoming" was originally applied only to this case,
but it is now applied more generally.
A host with embedded gateway functionality will
typically fall into the simple multihoming case. Note,
however, that a host may be simply multihomed without
containing an embedded gateway, i.e., without
forwarding datagrams from one connected network to
another.
This case presents the most difficult routing problems.
The choice of interface (i.e., the choice of first-hop
network) may significantly affect performance or even
reachability of remote parts of the Internet.
Finally, we note another possibility that is NOT
multihoming: one logical interface may be bound to multiple
physical interfaces, in order to increase the reliability or
throughput between directly connected machines by providing
alternative physical paths between them. For instance, two
systems might be connected by multiple point-to-point links.
We call this "link-layer multiplexing". With link-layer
multiplexing, the protocols above the link layer are unaware
that multiple physical interfaces are present; the link-
layer device driver is responsible for multiplexing and
routing packets across the physical interfaces.
In the Internet protocol architecture, a transport protocol
instance ("entity") has no address of its own, but instead
uses a single Internet Protocol (IP) address. This has
implications for the IP, transport, and application layers,
and for the interfaces between them. In particular, the
application software may have to be aware of the multiple IP
addresses of a multihomed host; in other cases, the choice
can be made within the network software.
3.3.4.2 Multihoming Requirements
The following general rules apply to the selection of an IP
source address for sending a datagram from a multihomed
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host.
(1) If the datagram is sent in response to a received
datagram, the source address for the response SHOULD be
the specific-destination address of the request. See
Sections 4.1.3.5 and 4.2.3.7 and the "General Issues"
section of [INTRO:1] for more specific requirements on
higher layers.
Otherwise, a source address must be selected.
(2) An application MUST be able to explicitly specify the
source address for initiating a connection or a
request.
(3) In the absence of such a specification, the networking
software MUST choose a source address. Rules for this
choice are described below.
There are two key requirement issues related to multihoming:
(A) A host MAY silently discard an incoming datagram whose
destination address does not correspond to the physical
interface through which it is received.
(B) A host MAY restrict itself to sending (non-source-
routed) IP datagrams only through the physical
interface that corresponds to the IP source address of
the datagrams.
DISCUSSION:
Internet host implementors have used two different
conceptual models for multihoming, briefly summarized
in the following discussion. This document takes no
stand on which model is preferred; each seems to have a
place. This ambivalence is reflected in the issues (A)
and (B) being optional.
o Strong ES Model
The Strong ES (End System, i.e., host) model
emphasizes the host/gateway (ES/IS) distinction,
and would therefore substitute MUST for MAY in
issues (A) and (B) above. It tends to model a
multihomed host as a set of logical hosts within
the same physical host.
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With respect to (A), proponents of the Strong ES
model note that automatic Internet routing
mechanisms could not route a datagram to a
physical interface that did not correspond to the
destination address.
Under the Strong ES model, the route computation
for an outgoing datagram is the mapping:
route(src IP addr, dest IP addr, TOS)
-> gateway
Here the source address is included as a parameter
in order to select a gateway that is directly
reachable on the corresponding physical interface.
Note that this model logically requires that in
general there be at least one default gateway, and
preferably multiple defaults, for each IP source
address.
o Weak ES Model
This view de-emphasizes the ES/IS distinction, and
would therefore substitute MUST NOT for MAY in
issues (A) and (B). This model may be the more
natural one for hosts that wiretap gateway routing
protocols, and is necessary for hosts that have
embedded gateway functionality.
The Weak ES Model may cause the Redirect mechanism
to fail. If a datagram is sent out a physical
interface that does not correspond to the
destination address, the first-hop gateway will
not realize when it needs to send a Redirect. On
the other hand, if the host has embedded gateway
functionality, then it has routing information
without listening to Redirects.
In the Weak ES model, the route computation for an
outgoing datagram is the mapping:
route(dest IP addr, TOS) -> gateway, interface
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3.3.4.3 Choosing a Source Address
DISCUSSION:
When it sends an initial connection request (e.g., a
TCP "SYN" segment) or a datagram service request (e.g.,
a UDP-based query), the transport layer on a multihomed
host needs to know which source address to use. If the
application does not specify it, the transport layer
must ask the IP layer to perform the conceptual
mapping:
GET_SRCADDR(remote IP addr, TOS)
-> local IP address
Here TOS is the Type-of-Service value (see Section
3.2.1.6), and the result is the desired source address.
The following rules are suggested for implementing this
mapping:
(a) If the remote Internet address lies on one of the
(sub-) nets to which the host is directly
connected, a corresponding source address may be
chosen, unless the corresponding interface is
known to be down.
(b) The route cache may be consulted, to see if there
is an active route to the specified destination
network through any network interface; if so, a
local IP address corresponding to that interface
may be chosen.
(c) The table of static routes, if any (see Section
3.3.1.2) may be similarly consulted.
(d) The default gateways may be consulted. If these
gateways are assigned to different interfaces, the
interface corresponding to the gateway with the
highest preference may be chosen.
In the future, there may be a defined way for a
multihomed host to ask the gateways on all connected
networks for advice about the best network to use for a
given destination.
IMPLEMENTATION:
It will be noted that this process is essentially the
same as datagram routing (see Section 3.3.1), and
therefore hosts may be able to combine the
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implementation of the two functions.
3.3.5 Source Route Forwarding
Subject to restrictions given below, a host MAY be able to act
as an intermediate hop in a source route, forwarding a source-
routed datagram to the next specified hop.
However, in performing this gateway-like function, the host
MUST obey all the relevant rules for a gateway forwarding
source-routed datagrams [INTRO:2]. This includes the following
specific provisions, which override the corresponding host
provisions given earlier in this document:
(A) TTL (ref. Section 3.2.1.7)
The TTL field MUST be decremented and the datagram perhaps
discarded as specified for a gateway in [INTRO:2].
(B) ICMP Destination Unreachable (ref. Section 3.2.2.1)
A host MUST be able to generate Destination Unreachable
messages with the following codes:
4 (Fragmentation Required but DF Set) when a source-
routed datagram cannot be fragmented to fit into the
target network;
5 (Source Route Failed) when a source-routed datagram
cannot be forwarded, e.g., because of a routing
problem or because the next hop of a strict source
route is not on a connected network.
(C) IP Source Address (ref. Section 3.2.1.3)
A source-routed datagram being forwarded MAY (and normally
will) have a source address that is not one of the IP
addresses of the forwarding host.
(D) Record Route Option (ref. Section 3.2.1.8d)
A host that is forwarding a source-routed datagram
containing a Record Route option MUST update that option,
if it has room.
(E) Timestamp Option (ref. Section 3.2.1.8e)
A host that is forwarding a source-routed datagram
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containing a Timestamp Option MUST add the current
timestamp to that option, according to the rules for this
option.
To define the rules restricting host forwarding of source-
routed datagrams, we use the term "local source-routing" if the
next hop will be through the same physical interface through
which the datagram arrived; otherwise, it is "non-local
source-routing".
o A host is permitted to perform local source-routing
without restriction.
o A host that supports non-local source-routing MUST have a
configurable switch to disable forwarding, and this switch
MUST default to disabled.
o The host MUST satisfy all gateway requirements for
configurable policy filters [INTRO:2] restricting non-
local forwarding.
If a host receives a datagram with an incomplete source route
but does not forward it for some reason, the host SHOULD return
an ICMP Destination Unreachable (code 5, Source Route Failed)
message, unless the datagram was itself an ICMP error message.
3.3.6 Broadcasts
Section 3.2.1.3 defined the four standard IP broadcast address
forms:
Limited Broadcast: {-1, -1}
Directed Broadcast: {,-1}
Subnet Directed Broadcast:
{,,-1}
All-Subnets Directed Broadcast: {,-1,-1}
A host MUST recognize any of these forms in the destination
address of an incoming datagram.
There is a class of hosts* that use non-standard broadcast
address forms, substituting 0 for -1. All hosts SHOULD
_________________________
*4.2BSD Unix and its derivatives, but not 4.3BSD.
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recognize and accept any of these non-standard broadcast
addresses as the destination address of an incoming datagram.
A host MAY optionally have a configuration option to choose the
0 or the -1 form of broadcast address, for each physical
interface, but this option SHOULD default to the standard (-1)
form.
When a host sends a datagram to a link-layer broadcast address,
the IP destination address MUST be a legal IP broadcast or IP
multicast address.
A host SHOULD silently discard a datagram that is received via
a link-layer broadcast (see Section 2.4) but does not specify
an IP multicast or broadcast destination address.
Hosts SHOULD use the Limited Broadcast address to broadcast to
a connected network.
DISCUSSION:
Using the Limited Broadcast address instead of a Directed
Broadcast address may improve system robustness. Problems
are often caused by machines that do not understand the
plethora of broadcast addresses (see Section 3.2.1.3), or
that may have different ideas about which broadcast
addresses are in use. The prime example of the latter is
machines that do not understand subnetting but are
attached to a subnetted net. Sending a Subnet Broadcast
for the connected network will confuse those machines,
which will see it as a message to some other host.
There has been discussion on whether a datagram addressed
to the Limited Broadcast address ought to be sent from all
the interfaces of a multihomed host. This specification
takes no stand on the issue.
3.3.7 IP Multicasting
A host SHOULD support local IP multicasting on all connected
networks for which a mapping from Class D IP addresses to
link-layer addresses has been specified (see below). Support
for local IP multicasting includes sending multicast datagrams,
joining multicast groups and receiving multicast datagrams, and
leaving multicast groups. This implies support for all of
[IP:4] except the IGMP protocol itself, which is OPTIONAL.
Internet Engineering Task Force [Page 67]
RFC1122 INTERNET LAYER October 1989
DISCUSSION:
IGMP provides gateways that are capable of multicast
routing with the information required to support IP
multicasting across multiple networks. At this time,
multicast-routing gateways are in the experimental stage
and are not widely available. For hosts that are not
connected to networks with multicast-routing gateways or
that do not need to receive multicast datagrams
originating on other networks, IGMP serves no purpose and
is therefore optional for now. However, the rest of
[IP:4] is currently recommended for the purpose of
providing IP-layer access to local network multicast
addressing, as a preferable alternative to local broadcast
addressing. It is expected that IGMP will become
recommended at some future date, when multicast-routing
gateways have become more widely available.
If IGMP is not implemented, a host SHOULD still join the "all-
hosts" group (224.0.0.1) when the IP layer is initialized and
remain a member for as long as the IP layer is active.
DISCUSSION:
Joining the "all-hosts" group will support strictly local
uses of multicasting, e.g., a gateway discovery protocol,
even if IGMP is not implemented.
The mapping of IP Class D addresses to local addresses is
currently specified for the following types of networks:
o Ethernet/IEEE 802.3, as defined in [IP:4].
o Any network that supports broadcast but not multicast,
addressing: all IP Class D addresses map to the local
broadcast address.
o Any type of point-to-point link (e.g., SLIP or HDLC
links): no mapping required. All IP multicast datagrams
are sent as-is, inside the local framing.
Mappings for other types of networks will be specified in the
future.
A host SHOULD provide a way for higher-layer protocols or
applications to determine which of the host's connected
network(s) support IP multicast addressing.
Internet Engineering Task Force [Page 68]
RFC1122 INTERNET LAYER October 1989
3.3.8 Error Reporting
Wherever practical, hosts MUST return ICMP error datagrams on
detection of an error, except in those cases where returning an
ICMP error message is specifically prohibited.
DISCUSSION:
A common phenomenon in datagram networks is the "black
hole disease": datagrams are sent out, but nothing comes
back. Without any error datagrams, it is difficult for
the user to figure out what the problem is.
3.4 INTERNET/TRANSPORT LAYER INTERFACE
The interface between the IP layer and the transport layer MUST
provide full access to all the mechanisms of the IP layer,
including options, Type-of-Service, and Time-to-Live. The
transport layer MUST either have mechanisms to set these interface
parameters, or provide a path to pass them through from an
application, or both.
DISCUSSION:
Applications are urged to make use of these mechanisms where
applicable, even when the mechanisms are not currently
effective in the Internet (e.g., TOS). This will allow these
mechanisms to be immediately useful when they do become
effective, without a large amount of retrofitting of host
software.
We now describe a conceptual interface between the transport layer
and the IP layer, as a set of procedure calls. This is an
extension of the information in Section 3.3 of RFC-791 [IP:1].
* Send Datagram
SEND(src, dst, prot, TOS, TTL, BufPTR, len, Id, DF, opt
=> result )
where the parameters are defined in RFC-791. Passing an Id
parameter is optional; see Section 3.2.1.5.
* Receive Datagram
RECV(BufPTR, prot
=> result, src, dst, SpecDest, TOS, len, opt)
Internet Engineering Task Force [Page 69]
RFC1122 INTERNET LAYER October 1989
All the parameters are defined in RFC-791, except for:
SpecDest = specific-destination address of datagram
(defined in Section 3.2.1.3)
The result parameter dst contains the datagram's destination
address. Since this may be a broadcast or multicast address,
the SpecDest parameter (not shown in RFC-791) MUST be passed.
The parameter opt contains all the IP options received in the
datagram; these MUST also be passed to the transport layer.
* Select Source Address
GET_SRCADDR(remote, TOS) -> local
remote = remote IP address
TOS = Type-of-Service
local = local IP address
See Section 3.3.4.3.
* Find Maximum Datagram Sizes
GET_MAXSIZES(local, remote, TOS) -> MMS_R, MMS_S
MMS_R = maximum receive transport-message size.
MMS_S = maximum send transport-message size.
(local, remote, TOS defined above)
See Sections 3.3.2 and 3.3.3.
* Advice on Delivery Success
ADVISE_DELIVPROB(sense, local, remote, TOS)
Here the parameter sense is a 1-bit flag indicating whether
positive or negative advice is being given; see the
discussion in Section 3.3.1.4. The other parameters were
defined earlier.
* Send ICMP Message
SEND_ICMP(src, dst, TOS, TTL, BufPTR, len, Id, DF, opt)
-> result
Internet Engineering Task Force [Page 70]
RFC1122 INTERNET LAYER October 1989
(Parameters defined in RFC-791).
Passing an Id parameter is optional; see Section 3.2.1.5.
The transport layer MUST be able to send certain ICMP
messages: Port Unreachable or any of the query-type
messages. This function could be considered to be a special
case of the SEND() call, of course; we describe it separately
for clarity.
* Receive ICMP Message
RECV_ICMP(BufPTR ) -> result, src, dst, len, opt
(Parameters defined in RFC-791).
The IP layer MUST pass certain ICMP messages up to the
appropriate transport-layer routine. This function could be
considered to be a special case of the RECV() call, of
course; we describe it separately for clarity.
For an ICMP error message, the data that is passed up MUST
include the original Internet header plus all the octets of
the original message that are included in the ICMP message.
This data will be used by the transport layer to locate the
connection state information, if any.
In particular, the following ICMP messages are to be passed
up:
o Destination Unreachable
o Source Quench
o Echo Reply (to ICMP user interface, unless the Echo
Request originated in the IP layer)
o Timestamp Reply (to ICMP user interface)
o Time Exceeded
DISCUSSION:
In the future, there may be additions to this interface to
pass path data (see Section 3.3.1.3) between the IP and
transport layers.
Internet Engineering Task Force [Page 71]
RFC1122 INTERNET LAYER October 1989
3.5 INTERNET LAYER REQUIREMENTS SUMMARY
| | | | |S| |
| | | | |H| |F
| | | | |O|M|o
| | |S| |U|U|o
| | |H| |L|S|t
| |M|O| |D|T|n
| |U|U|M| | |o
| |S|L|A|N|N|t
| |T|D|Y|O|O|t
FEATURE |SECTION | | | |T|T|e
-------------------------------------------------|--------|-|-|-|-|-|--
| | | | | | |
Implement IP and ICMP |3.1 |x| | | | |
Handle remote multihoming in application layer |3.1 |x| | | | |
Support local multihoming |3.1 | | |x| | |
Meet gateway specs if forward datagrams |3.1 |x| | | | |
Configuration switch for embedded gateway |3.1 |x| | | | |1
Config switch default to non-gateway |3.1 |x| | | | |1
Auto-config based on number of interfaces |3.1 | | | | |x|1
Able to log discarded datagrams |3.1 | |x| | | |
Record in counter |3.1 | |x| | | |
| | | | | | |
Silently discard Version != 4 |3.2.1.1 |x| | | | |
Verify IP checksum, silently discard bad dgram |3.2.1.2 |x| | | | |
Addressing: | | | | | | |
Subnet addressing (RFC-950) |3.2.1.3 |x| | | | |
Src address must be host's own IP address |3.2.1.3 |x| | | | |
Silently discard datagram with bad dest addr |3.2.1.3 |x| | | | |
Silently discard datagram with bad src addr |3.2.1.3 |x| | | | |
Support reassembly |3.2.1.4 |x| | | | |
Retain same Id field in identical datagram |3.2.1.5 | | |x| | |
| | | | | | |
TOS: | | | | | | |
Allow transport layer to set TOS |3.2.1.6 |x| | | | |
Pass received TOS up to transport layer |3.2.1.6 | |x| | | |
Use RFC-795 link-layer mappings for TOS |3.2.1.6 | | | |x| |
TTL: | | | | | | |
Send packet with TTL of 0 |3.2.1.7 | | | | |x|
Discard received packets with TTL < 2 |3.2.1.7 | | | | |x|
Allow transport layer to set TTL |3.2.1.7 |x| | | | |
Fixed TTL is configurable |3.2.1.7 |x| | | | |
| | | | | | |
IP Options: | | | | | | |
Allow transport layer to send IP options |3.2.1.8 |x| | | | |
Pass all IP options rcvd to higher layer |3.2.1.8 |x| | | | |
Internet Engineering Task Force [Page 72]
RFC1122 INTERNET LAYER October 1989
IP layer silently ignore unknown options |3.2.1.8 |x| | | | |
Security option |3.2.1.8a| | |x| | |
Send Stream Identifier option |3.2.1.8b| | | |x| |
Silently ignore Stream Identifer option |3.2.1.8b|x| | | | |
Record Route option |3.2.1.8d| | |x| | |
Timestamp option |3.2.1.8e| | |x| | |
Source Route Option: | | | | | | |
Originate & terminate Source Route options |3.2.1.8c|x| | | | |
Datagram with completed SR passed up to TL |3.2.1.8c|x| | | | |
Build correct (non-redundant) return route |3.2.1.8c|x| | | | |
Send multiple SR options in one header |3.2.1.8c| | | | |x|
| | | | | | |
ICMP: | | | | | | |
Silently discard ICMP msg with unknown type |3.2.2 |x| | | | |
Include more than 8 octets of orig datagram |3.2.2 | | |x| | |
Included octets same as received |3.2.2 |x| | | | |
Demux ICMP Error to transport protocol |3.2.2 |x| | | | |
Send ICMP error message with TOS=0 |3.2.2 | |x| | | |
Send ICMP error message for: | | | | | | |
- ICMP error msg |3.2.2 | | | | |x|
- IP b'cast or IP m'cast |3.2.2 | | | | |x|
- Link-layer b'cast |3.2.2 | | | | |x|
- Non-initial fragment |3.2.2 | | | | |x|
- Datagram with non-unique src address |3.2.2 | | | | |x|
Return ICMP error msgs (when not prohibited) |3.3.8 |x| | | | |
| | | | | | |
Dest Unreachable: | | | | | | |
Generate Dest Unreachable (code 2/3) |3.2.2.1 | |x| | | |
Pass ICMP Dest Unreachable to higher layer |3.2.2.1 |x| | | | |
Higher layer act on Dest Unreach |3.2.2.1 | |x| | | |
Interpret Dest Unreach as only hint |3.2.2.1 |x| | | | |
Redirect: | | | | | | |
Host send Redirect |3.2.2.2 | | | |x| |
Update route cache when recv Redirect |3.2.2.2 |x| | | | |
Handle both Host and Net Redirects |3.2.2.2 |x| | | | |
Discard illegal Redirect |3.2.2.2 | |x| | | |
Source Quench: | | | | | | |
Send Source Quench if buffering exceeded |3.2.2.3 | | |x| | |
Pass Source Quench to higher layer |3.2.2.3 |x| | | | |
Higher layer act on Source Quench |3.2.2.3 | |x| | | |
Time Exceeded: pass to higher layer |3.2.2.4 |x| | | | |
Parameter Problem: | | | | | | |
Send Parameter Problem messages |3.2.2.5 | |x| | | |
Pass Parameter Problem to higher layer |3.2.2.5 |x| | | | |
Report Parameter Problem to user |3.2.2.5 | | |x| | |
| | | | | | |
ICMP Echo Request or Reply: | | | | | | |
Echo server and Echo client |3.2.2.6 |x| | | | |
Internet Engineering Task Force [Page 73]
RFC1122 INTERNET LAYER October 1989
Echo client |3.2.2.6 | |x| | | |
Discard Echo Request to broadcast address |3.2.2.6 | | |x| | |
Discard Echo Request to multicast address |3.2.2.6 | | |x| | |
Use specific-dest addr as Echo Reply src |3.2.2.6 |x| | | | |
Send same data in Echo Reply |3.2.2.6 |x| | | | |
Pass Echo Reply to higher layer |3.2.2.6 |x| | | | |
Reflect Record Route, Time Stamp options |3.2.2.6 | |x| | | |
Reverse and reflect Source Route option |3.2.2.6 |x| | | | |
| | | | | | |
ICMP Information Request or Reply: |3.2.2.7 | | | |x| |
ICMP Timestamp and Timestamp Reply: |3.2.2.8 | | |x| | |
Minimize delay variability |3.2.2.8 | |x| | | |1
Silently discard b'cast Timestamp |3.2.2.8 | | |x| | |1
Silently discard m'cast Timestamp |3.2.2.8 | | |x| | |1
Use specific-dest addr as TS Reply src |3.2.2.8 |x| | | | |1
Reflect Record Route, Time Stamp options |3.2.2.6 | |x| | | |1
Reverse and reflect Source Route option |3.2.2.8 |x| | | | |1
Pass Timestamp Reply to higher layer |3.2.2.8 |x| | | | |1
Obey rules for "standard value" |3.2.2.8 |x| | | | |1
| | | | | | |
ICMP Address Mask Request and Reply: | | | | | | |
Addr Mask source configurable |3.2.2.9 |x| | | | |
Support static configuration of addr mask |3.2.2.9 |x| | | | |
Get addr mask dynamically during booting |3.2.2.9 | | |x| | |
Get addr via ICMP Addr Mask Request/Reply |3.2.2.9 | | |x| | |
Retransmit Addr Mask Req if no Reply |3.2.2.9 |x| | | | |3
Assume default mask if no Reply |3.2.2.9 | |x| | | |3
Update address mask from first Reply only |3.2.2.9 |x| | | | |3
Reasonableness check on Addr Mask |3.2.2.9 | |x| | | |
Send unauthorized Addr Mask Reply msgs |3.2.2.9 | | | | |x|
Explicitly configured to be agent |3.2.2.9 |x| | | | |
Static config=> Addr-Mask-Authoritative flag |3.2.2.9 | |x| | | |
Broadcast Addr Mask Reply when init. |3.2.2.9 |x| | | | |3
| | | | | | |
ROUTING OUTBOUND DATAGRAMS: | | | | | | |
Use address mask in local/remote decision |3.3.1.1 |x| | | | |
Operate with no gateways on conn network |3.3.1.1 |x| | | | |
Maintain "route cache" of next-hop gateways |3.3.1.2 |x| | | | |
Treat Host and Net Redirect the same |3.3.1.2 | |x| | | |
If no cache entry, use default gateway |3.3.1.2 |x| | | | |
Support multiple default gateways |3.3.1.2 |x| | | | |
Provide table of static routes |3.3.1.2 | | |x| | |
Flag: route overridable by Redirects |3.3.1.2 | | |x| | |
Key route cache on host, not net address |3.3.1.3 | | |x| | |
Include TOS in route cache |3.3.1.3 | |x| | | |
| | | | | | |
Able to detect failure of next-hop gateway |3.3.1.4 |x| | | | |
Assume route is good forever |3.3.1.4 | | | |x| |
Internet Engineering Task Force [Page 74]
RFC1122 INTERNET LAYER October 1989
Ping gateways continuously |3.3.1.4 | | | | |x|
Ping only when traffic being sent |3.3.1.4 |x| | | | |
Ping only when no positive indication |3.3.1.4 |x| | | | |
Higher and lower layers give advice |3.3.1.4 | |x| | | |
Switch from failed default g'way to another |3.3.1.5 |x| | | | |
Manual method of entering config info |3.3.1.6 |x| | | | |
| | | | | | |
REASSEMBLY and FRAGMENTATION: | | | | | | |
Able to reassemble incoming datagrams |3.3.2 |x| | | | |
At least 576 byte datagrams |3.3.2 |x| | | | |
EMTU_R configurable or indefinite |3.3.2 | |x| | | |
Transport layer able to learn MMS_R |3.3.2 |x| | | | |
Send ICMP Time Exceeded on reassembly timeout |3.3.2 |x| | | | |
Fixed reassembly timeout value |3.3.2 | |x| | | |
| | | | | | |
Pass MMS_S to higher layers |3.3.3 |x| | | | |
Local fragmentation of outgoing packets |3.3.3 | | |x| | |
Else don't send bigger than MMS_S |3.3.3 |x| | | | |
Send max 576 to off-net destination |3.3.3 | |x| | | |
All-Subnets-MTU configuration flag |3.3.3 | | |x| | |
| | | | | | |
MULTIHOMING: | | | | | | |
Reply with same addr as spec-dest addr |3.3.4.2 | |x| | | |
Allow application to choose local IP addr |3.3.4.2 |x| | | | |
Silently discard d'gram in "wrong" interface |3.3.4.2 | | |x| | |
Only send d'gram through "right" interface |3.3.4.2 | | |x| | |4
| | | | | | |
SOURCE-ROUTE FORWARDING: | | | | | | |
Forward datagram with Source Route option |3.3.5 | | |x| | |1
Obey corresponding gateway rules |3.3.5 |x| | | | |1
Update TTL by gateway rules |3.3.5 |x| | | | |1
Able to generate ICMP err code 4, 5 |3.3.5 |x| | | | |1
IP src addr not local host |3.3.5 | | |x| | |1
Update Timestamp, Record Route options |3.3.5 |x| | | | |1
Configurable switch for non-local SRing |3.3.5 |x| | | | |1
Defaults to OFF |3.3.5 |x| | | | |1
Satisfy gwy access rules for non-local SRing |3.3.5 |x| | | | |1
If not forward, send Dest Unreach (cd 5) |3.3.5 | |x| | | |2
| | | | | | |
BROADCAST: | | | | | | |
Broadcast addr as IP source addr |3.2.1.3 | | | | |x|
Receive 0 or -1 broadcast formats OK |3.3.6 | |x| | | |
Config'ble option to send 0 or -1 b'cast |3.3.6 | | |x| | |
Default to -1 broadcast |3.3.6 | |x| | | |
Recognize all broadcast address formats |3.3.6 |x| | | | |
Use IP b'cast/m'cast addr in link-layer b'cast |3.3.6 |x| | | | |
Silently discard link-layer-only b'cast dg's |3.3.6 | |x| | | |
Use Limited Broadcast addr for connected net |3.3.6 | |x| | | |
Internet Engineering Task Force [Page 75]
RFC1122 INTERNET LAYER October 1989
| | | | | | |
MULTICAST: | | | | | | |
Support local IP multicasting (RFC-1112) |3.3.7 | |x| | | |
Support IGMP (RFC-1112) |3.3.7 | | |x| | |
Join all-hosts group at startup |3.3.7 | |x| | | |
Higher layers learn i'face m'cast capability |3.3.7 | |x| | | |
| | | | | | |
INTERFACE: | | | | | | |
Allow transport layer to use all IP mechanisms |3.4 |x| | | | |
Pass interface ident up to transport layer |3.4 |x| | | | |
Pass all IP options up to transport layer |3.4 |x| | | | |
Transport layer can send certain ICMP messages |3.4 |x| | | | |
Pass spec'd ICMP messages up to transp. layer |3.4 |x| | | | |
Include IP hdr+8 octets or more from orig. |3.4 |x| | | | |
Able to leap tall buildings at a single bound |3.5 | |x| | | |
Footnotes:
(1) Only if feature is implemented.
(2) This requirement is overruled if datagram is an ICMP error message.
(3) Only if feature is implemented and is configured "on".
(4) Unless has embedded gateway functionality or is source routed.
Internet Engineering Task Force [Page 76]
RFC1122 TRANSPORT LAYER -- UDP October 1989
4. TRANSPORT PROTOCOLS
4.1 USER DATAGRAM PROTOCOL -- UDP
4.1.1 INTRODUCTION
The User Datagram Protocol UDP [UDP:1] offers only a minimal
transport service -- non-guaranteed datagram delivery -- and
gives applications direct access to the datagram service of the
IP layer. UDP is used by applications that do not require the
level of service of TCP or that wish to use communications
services (e.g., multicast or broadcast delivery) not available
from TCP.
UDP is almost a null protocol; the only services it provides
over IP are checksumming of data and multiplexing by port
number. Therefore, an application program running over UDP
must deal directly with end-to-end communication problems that
a connection-oriented protocol would have handled -- e.g.,
retransmission for reliable delivery, packetization and
reassembly, flow control, congestion avoidance, etc., when
these are required. The fairly complex coupling between IP and
TCP will be mirrored in the coupling between UDP and many
applications using UDP.
4.1.2 PROTOCOL WALK-THROUGH
There are no known errors in the specification of UDP.
4.1.3 SPECIFIC ISSUES
4.1.3.1 Ports
UDP well-known ports follow the same rules as TCP well-known
ports; see Section 4.2.2.1 below.
If a datagram arrives addressed to a UDP port for which
there is no pending LISTEN call, UDP SHOULD send an ICMP
Port Unreachable message.
4.1.3.2 IP Options
UDP MUST pass any IP option that it receives from the IP
layer transparently to the application layer.
An application MUST be able to specify IP options to be sent
in its UDP datagrams, and UDP MUST pass these options to the
IP layer.
Internet Engineering Task Force [Page 77]
RFC1122 TRANSPORT LAYER -- UDP October 1989
DISCUSSION:
At present, the only options that need be passed
through UDP are Source Route, Record Route, and Time
Stamp. However, new options may be defined in the
future, and UDP need not and should not make any
assumptions about the format or content of options it
passes to or from the application; an exception to this
might be an IP-layer security option.
An application based on UDP will need to obtain a
source route from a request datagram and supply a
reversed route for sending the corresponding reply.
4.1.3.3 ICMP Messages
UDP MUST pass to the application layer all ICMP error
messages that it receives from the IP layer. Conceptually
at least, this may be accomplished with an upcall to the
ERROR_REPORT routine (see Section 4.2.4.1).
DISCUSSION:
Note that ICMP error messages resulting from sending a
UDP datagram are received asynchronously. A UDP-based
application that wants to receive ICMP error messages
is responsible for maintaining the state necessary to
demultiplex these messages when they arrive; for
example, the application may keep a pending receive
operation for this purpose. The application is also
responsible to avoid confusion from a delayed ICMP
error message resulting from an earlier use of the same
port(s).
4.1.3.4 UDP Checksums
A host MUST implement the facility to generate and validate
UDP checksums. An application MAY optionally be able to
control whether a UDP checksum will be generated, but it
MUST default to checksumming on.
If a UDP datagram is received with a checksum that is non-
zero and invalid, UDP MUST silently discard the datagram.
An application MAY optionally be able to control whether UDP
datagrams without checksums should be discarded or passed to
the application.
DISCUSSION:
Some applications that normally run only across local
area networks have chosen to turn off UDP checksums for
Internet Engineering Task Force [Page 78]
RFC1122 TRANSPORT LAYER -- UDP October 1989
efficiency. As a result, numerous cases of undetected
errors have been reported. The advisability of ever
turning off UDP checksumming is very controversial.
IMPLEMENTATION:
There is a common implementation error in UDP
checksums. Unlike the TCP checksum, the UDP checksum
is optional; the value zero is transmitted in the
checksum field of a UDP header to indicate the absence
of a checksum. If the transmitter really calculates a
UDP checksum of zero, it must transmit the checksum as
all 1's (65535). No special action is required at the
receiver, since zero and 65535 are equivalent in 1's
complement arithmetic.
4.1.3.5 UDP Multihoming
When a UDP datagram is received, its specific-destination
address MUST be passed up to the application layer.
An application program MUST be able to specify the IP source
address to be used for sending a UDP datagram or to leave it
unspecified (in which case the networking software will
choose an appropriate source address). There SHOULD be a
way to communicate the chosen source address up to the
application layer (e.g, so that the application can later
receive a reply datagram only from the corresponding
interface).
DISCUSSION:
A request/response application that uses UDP should use
a source address for the response that is the same as
the specific destination address of the request. See
the "General Issues" section of [INTRO:1].
4.1.3.6 Invalid Addresses
A UDP datagram received with an invalid IP source address
(e.g., a broadcast or multicast address) must be discarded
by UDP or by the IP layer (see Section 3.2.1.3).
When a host sends a UDP datagram, the source address MUST be
(one of) the IP address(es) of the host.
4.1.4 UDP/APPLICATION LAYER INTERFACE
The application interface to UDP MUST provide the full services
of the IP/transport interface described in Section 3.4 of this
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document. Thus, an application using UDP needs the functions
of the GET_SRCADDR(), GET_MAXSIZES(), ADVISE_DELIVPROB(), and
RECV_ICMP() calls described in Section 3.4. For example,
GET_MAXSIZES() can be used to learn the effective maximum UDP
maximum datagram size for a particular {interface,remote
host,TOS} triplet.
An application-layer program MUST be able to set the TTL and
TOS values as well as IP options for sending a UDP datagram,
and these values must be passed transparently to the IP layer.
UDP MAY pass the received TOS up to the application layer.
4.1.5 UDP REQUIREMENTS SUMMARY
| | | | |S| |
| | | | |H| |F
| | | | |O|M|o
| | |S| |U|U|o
| | |H| |L|S|t
| |M|O| |D|T|n
| |U|U|M| | |o
| |S|L|A|N|N|t
| |T|D|Y|O|O|t
FEATURE |SECTION | | | |T|T|e
-------------------------------------------------|--------|-|-|-|-|-|--
| | | | | | |
UDP | | | | | | |
-------------------------------------------------|--------|-|-|-|-|-|--
| | | | | | |
UDP send Port Unreachable |4.1.3.1 | |x| | | |
| | | | | | |
IP Options in UDP | | | | | | |
- Pass rcv'd IP options to applic layer |4.1.3.2 |x| | | | |
- Applic layer can specify IP options in Send |4.1.3.2 |x| | | | |
- UDP passes IP options down to IP layer |4.1.3.2 |x| | | | |
| | | | | | |
Pass ICMP msgs up to applic layer |4.1.3.3 |x| | | | |
| | | | | | |
UDP checksums: | | | | | | |
- Able to generate/check checksum |4.1.3.4 |x| | | | |
- Silently discard bad checksum |4.1.3.4 |x| | | | |
- Sender Option to not generate checksum |4.1.3.4 | | |x| | |
- Default is to checksum |4.1.3.4 |x| | | | |
- Receiver Option to require checksum |4.1.3.4 | | |x| | |
| | | | | | |
UDP Multihoming | | | | | | |
- Pass spec-dest addr to application |4.1.3.5 |x| | | | |
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- Applic layer can specify Local IP addr |4.1.3.5 |x| | | | |
- Applic layer specify wild Local IP addr |4.1.3.5 |x| | | | |
- Applic layer notified of Local IP addr used |4.1.3.5 | |x| | | |
| | | | | | |
Bad IP src addr silently discarded by UDP/IP |4.1.3.6 |x| | | | |
Only send valid IP source address |4.1.3.6 |x| | | | |
UDP Application Interface Services | | | | | | |
Full IP interface of 3.4 for application |4.1.4 |x| | | | |
- Able to spec TTL, TOS, IP opts when send dg |4.1.4 |x| | | | |
- Pass received TOS up to applic layer |4.1.4 | | |x| | |
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4.2 TRANSMISSION CONTROL PROTOCOL -- TCP
4.2.1 INTRODUCTION
The Transmission Control Protocol TCP [TCP:1] is the primary
virtual-circuit transport protocol for the Internet suite. TCP
provides reliable, in-sequence delivery of a full-duplex stream
of octets (8-bit bytes). TCP is used by those applications
needing reliable, connection-oriented transport service, e.g.,
mail (SMTP), file transfer (FTP), and virtual terminal service
(Telnet); requirements for these application-layer protocols
are described in [INTRO:1].
4.2.2 PROTOCOL WALK-THROUGH
4.2.2.1 Well-Known Ports: RFC-793 Section 2.7
DISCUSSION:
TCP reserves port numbers in the range 0-255 for
"well-known" ports, used to access services that are
standardized across the Internet. The remainder of the
port space can be freely allocated to application
processes. Current well-known port definitions are
listed in the RFC entitled "Assigned Numbers"
[INTRO:6]. A prerequisite for defining a new well-
known port is an RFC documenting the proposed service
in enough detail to allow new implementations.
Some systems extend this notion by adding a third
subdivision of the TCP port space: reserved ports,
which are generally used for operating-system-specific
services. For example, reserved ports might fall
between 256 and some system-dependent upper limit.
Some systems further choose to protect well-known and
reserved ports by permitting only privileged users to
open TCP connections with those port values. This is
perfectly reasonable as long as the host does not
assume that all hosts protect their low-numbered ports
in this manner.
4.2.2.2 Use of Push: RFC-793 Section 2.8
When an application issues a series of SEND calls without
setting the PUSH flag, the TCP MAY aggregate the data
internally without sending it. Similarly, when a series of
segments is received without the PSH bit, a TCP MAY queue
the data internally without passing it to the receiving
application.
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The PSH bit is not a record marker and is independent of
segment boundaries. The transmitter SHOULD collapse
successive PSH bits when it packetizes data, to send the
largest possible segment.
A TCP MAY implement PUSH flags on SEND calls. If PUSH flags
are not implemented, then the sending TCP: (1) must not
buffer data indefinitely, and (2) MUST set the PSH bit in
the last buffered segment (i.e., when there is no more
queued data to be sent).
The discussion in RFC-793 on pages 48, 50, and 74
erroneously implies that a received PSH flag must be passed
to the application layer. Passing a received PSH flag to
the application layer is now OPTIONAL.
An application program is logically required to set the PUSH
flag in a SEND call whenever it needs to force delivery of
the data to avoid a communication deadlock. However, a TCP
SHOULD send a maximum-sized segment whenever possible, to
improve performance (see Section 4.2.3.4).
DISCUSSION:
When the PUSH flag is not implemented on SEND calls,
i.e., when the application/TCP interface uses a pure
streaming model, responsibility for aggregating any
tiny data fragments to form reasonable sized segments
is partially borne by the application layer.
Generally, an interactive application protocol must set
the PUSH flag at least in the last SEND call in each
command or response sequence. A bulk transfer protocol
like FTP should set the PUSH flag on the last segment
of a file or when necessary to prevent buffer deadlock.
At the receiver, the PSH bit forces buffered data to be
delivered to the application (even if less than a full
buffer has been received). Conversely, the lack of a
PSH bit can be used to avoid unnecessary wakeup calls
to the application process; this can be an important
performance optimization for large timesharing hosts.
Passing the PSH bit to the receiving application allows
an analogous optimization within the application.
4.2.2.3 Window Size: RFC-793 Section 3.1
The window size MUST be treated as an unsigned number, or
else large window sizes will appear like negative windows
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and TCP will not work. It is RECOMMENDED that
implementations reserve 32-bit fields for the send and
receive window sizes in the connection record and do all
window computations with 32 bits.
DISCUSSION:
It is known that the window field in the TCP header is
too small for high-speed, long-delay paths.
Experimental TCP options have been defined to extend
the window size; see for example [TCP:11]. In
anticipation of the adoption of such an extension, TCP
implementors should treat windows as 32 bits.
4.2.2.4 Urgent Pointer: RFC-793 Section 3.1
The second sentence is in error: the urgent pointer points
to the sequence number of the LAST octet (not LAST+1) in a
sequence of urgent data. The description on page 56 (last
sentence) is correct.
A TCP MUST support a sequence of urgent data of any length.
A TCP MUST inform the application layer asynchronously
whenever it receives an Urgent pointer and there was
previously no pending urgent data, or whenever the Urgent
pointer advances in the data stream. There MUST be a way
for the application to learn how much urgent data remains to
be read from the connection, or at least to determine
whether or not more urgent data remains to be read.
DISCUSSION:
Although the Urgent mechanism may be used for any
application, it is normally used to send "interrupt"-
type commands to a Telnet program (see "Using Telnet
Synch Sequence" section in [INTRO:1]).
The asynchronous or "out-of-band" notification will
allow the application to go into "urgent mode", reading
data from the TCP connection. This allows control
commands to be sent to an application whose normal
input buffers are full of unprocessed data.
IMPLEMENTATION:
The generic ERROR-REPORT() upcall described in Section
4.2.4.1 is a possible mechanism for informing the
application of the arrival of urgent data.
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4.2.2.5 TCP Options: RFC-793 Section 3.1
A TCP MUST be able to receive a TCP option in any segment.
A TCP MUST ignore without error any TCP option it does not
implement, assuming that the option has a length field (all
TCP options defined in the future will have length fields).
TCP MUST be prepared to handle an illegal option length
(e.g., zero) without crashing; a suggested procedure is to
reset the connection and log the reason.
4.2.2.6 Maximum Segment Size Option: RFC-793 Section 3.1
TCP MUST implement both sending and receiving the Maximum
Segment Size option [TCP:4].
TCP SHOULD send an MSS (Maximum Segment Size) option in
every SYN segment when its receive MSS differs from the
default 536, and MAY send it always.
If an MSS option is not received at connection setup, TCP
MUST assume a default send MSS of 536 (576-40) [TCP:4].
The maximum size of a segment that TCP really sends, the
"effective send MSS," MUST be the smaller of the send MSS
(which reflects the available reassembly buffer size at the
remote host) and the largest size permitted by the IP layer:
Eff.snd.MSS =
min(SendMSS+20, MMS_S) - TCPhdrsize - IPoptionsize
where:
* SendMSS is the MSS value received from the remote host,
or the default 536 if no MSS option is received.
* MMS_S is the maximum size for a transport-layer message
that TCP may send.
* TCPhdrsize is the size of the TCP header; this is
normally 20, but may be larger if TCP options are to be
sent.
* IPoptionsize is the size of any IP options that TCP
will pass to the IP layer with the current message.
The MSS value to be sent in an MSS option must be less than
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or equal to:
MMS_R - 20
where MMS_R is the maximum size for a transport-layer
message that can be received (and reassembled). TCP obtains
MMS_R and MMS_S from the IP layer; see the generic call
GET_MAXSIZES in Section 3.4.
DISCUSSION:
The choice of TCP segment size has a strong effect on
performance. Larger segments increase throughput by
amortizing header size and per-datagram processing
overhead over more data bytes; however, if the packet
is so large that it causes IP fragmentation, efficiency
drops sharply if any fragments are lost [IP:9].
Some TCP implementations send an MSS option only if the
destination host is on a non-connected network.
However, in general the TCP layer may not have the
appropriate information to make this decision, so it is
preferable to leave to the IP layer the task of
determining a suitable MTU for the Internet path. We
therefore recommend that TCP always send the option (if
not 536) and that the IP layer determine MMS_R as
specified in 3.3.3 and 3.4. A proposed IP-layer
mechanism to measure the MTU would then modify the IP
layer without changing TCP.
4.2.2.7 TCP Checksum: RFC-793 Section 3.1
Unlike the UDP checksum (see Section 4.1.3.4), the TCP
checksum is never optional. The sender MUST generate it and
the receiver MUST check it.
4.2.2.8 TCP Connection State Diagram: RFC-793 Section 3.2,
page 23
There are several problems with this diagram:
(a) The arrow from SYN-SENT to SYN-RCVD should be labeled
with "snd SYN,ACK", to agree with the text on page 68
and with Figure 8.
(b) There could be an arrow from SYN-RCVD state to LISTEN
state, conditioned on receiving a RST after a passive
open (see text page 70).
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(c) It is possible to go directly from FIN-WAIT-1 to the
TIME-WAIT state (see page 75 of the spec).
4.2.2.9 Initial Sequence Number Selection: RFC-793 Section
3.3, page 27
A TCP MUST use the specified clock-driven selection of
initial sequence numbers.
4.2.2.10 Simultaneous Open Attempts: RFC-793 Section 3.4, page
32
There is an error in Figure 8: the packet on line 7 should
be identical to the packet on line 5.
A TCP MUST support simultaneous open attempts.
DISCUSSION:
It sometimes surprises implementors that if two
applications attempt to simultaneously connect to each
other, only one connection is generated instead of two.
This was an intentional design decision; don't try to
"fix" it.
4.2.2.11 Recovery from Old Duplicate SYN: RFC-793 Section 3.4,
page 33
Note that a TCP implementation MUST keep track of whether a
connection has reached SYN_RCVD state as the result of a
passive OPEN or an active OPEN.
4.2.2.12 RST Segment: RFC-793 Section 3.4
A TCP SHOULD allow a received RST segment to include data.
DISCUSSION
It has been suggested that a RST segment could contain
ASCII text that encoded and explained the cause of the
RST. No standard has yet been established for such
data.
4.2.2.13 Closing a Connection: RFC-793 Section 3.5
A TCP connection may terminate in two ways: (1) the normal
TCP close sequence using a FIN handshake, and (2) an "abort"
in which one or more RST segments are sent and the
connection state is immediately discarded. If a TCP
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connection is closed by the remote site, the local
application MUST be informed whether it closed normally or
was aborted.
The normal TCP close sequence delivers buffered data
reliably in both directions. Since the two directions of a
TCP connection are closed independently, it is possible for
a connection to be "half closed," i.e., closed in only one
direction, and a host is permitted to continue sending data
in the open direction on a half-closed connection.
A host MAY implement a "half-duplex" TCP close sequence, so
that an application that has called CLOSE cannot continue to
read data from the connection. If such a host issues a
CLOSE call while received data is still pending in TCP, or
if new data is received after CLOSE is called, its TCP
SHOULD send a RST to show that data was lost.
When a connection is closed actively, it MUST linger in
TIME-WAIT state for a time 2xMSL (Maximum Segment Lifetime).
However, it MAY accept a new SYN from the remote TCP to
reopen the connection directly from TIME-WAIT state, if it:
(1) assigns its initial sequence number for the new
connection to be larger than the largest sequence
number it used on the previous connection incarnation,
and
(2) returns to TIME-WAIT state if the SYN turns out to be
an old duplicate.
DISCUSSION:
TCP's full-duplex data-preserving close is a feature
that is not included in the analogous ISO transport
protocol TP4.
Some systems have not implemented half-closed
connections, presumably because they do not fit into
the I/O model of their particular operating system. On
these systems, once an application has called CLOSE, it
can no longer read input data from the connection; this
is referred to as a "half-duplex" TCP close sequence.
The graceful close algorithm of TCP requires that the
connection state remain defined on (at least) one end
of the connection, for a timeout period of 2xMSL, i.e.,
4 minutes. During this period, the (remote socket,
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local socket) pair that defines the connection is busy
and cannot be reused. To shorten the time that a given
port pair is tied up, some TCPs allow a new SYN to be
accepted in TIME-WAIT state.
4.2.2.14 Data Communication: RFC-793 Section 3.7, page 40
Since RFC-793 was written, there has been extensive work on
TCP algorithms to achieve efficient data communication.
Later sections of the present document describe required and
recommended TCP algorithms to determine when to send data
(Section 4.2.3.4), when to send an acknowledgment (Section
4.2.3.2), and when to update the window (Section 4.2.3.3).
DISCUSSION:
One important performance issue is "Silly Window
Syndrome" or "SWS" [TCP:5], a stable pattern of small
incremental window movements resulting in extremely
poor TCP performance. Algorithms to avoid SWS are
described below for both the sending side (Section
4.2.3.4) and the receiving side (Section 4.2.3.3).
In brief, SWS is caused by the receiver advancing the
right window edge whenever it has any new buffer space
available to receive data and by the sender using any
incremental window, no matter how small, to send more
data [TCP:5]. The result can be a stable pattern of
sending tiny data segments, even though both sender and
receiver have a large total buffer space for the
connection. SWS can only occur during the transmission
of a large amount of data; if the connection goes
quiescent, the problem will disappear. It is caused by
typical straightforward implementation of window
management, but the sender and receiver algorithms
given below will avoid it.
Another important TCP performance issue is that some
applications, especially remote login to character-at-
a-time hosts, tend to send streams of one-octet data
segments. To avoid deadlocks, every TCP SEND call from
such applications must be "pushed", either explicitly
by the application or else implicitly by TCP. The
result may be a stream of TCP segments that contain one
data octet each, which makes very inefficient use of
the Internet and contributes to Internet congestion.
The Nagle Algorithm described in Section 4.2.3.4
provides a simple and effective solution to this
problem. It does have the effect of clumping
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characters over Telnet connections; this may initially
surprise users accustomed to single-character echo, but
user acceptance has not been a problem.
Note that the Nagle algorithm and the send SWS
avoidance algorithm play complementary roles in
improving performance. The Nagle algorithm discourages
sending tiny segments when the data to be sent
increases in small increments, while the SWS avoidance
algorithm discourages small segments resulting from the
right window edge advancing in small increments.
A careless implementation can send two or more
acknowledgment segments per data segment received. For
example, suppose the receiver acknowledges every data
segment immediately. When the application program
subsequently consumes the data and increases the
available receive buffer space again, the receiver may
send a second acknowledgment segment to update the
window at the sender. The extreme case occurs with
single-character segments on TCP connections using the
Telnet protocol for remote login service. Some
implementations have been observed in which each
incoming 1-character segment generates three return
segments: (1) the acknowledgment, (2) a one byte
increase in the window, and (3) the echoed character,
respectively.
4.2.2.15 Retransmission Timeout: RFC-793 Section 3.7, page 41
The algorithm suggested in RFC-793 for calculating the
retransmission timeout is now known to be inadequate; see
Section 4.2.3.1 below.
Recent work by Jacobson [TCP:7] on Internet congestion and
TCP retransmission stability has produced a transmission
algorithm combining "slow start" with "congestion
avoidance". A TCP MUST implement this algorithm.
If a retransmitted packet is identical to the original
packet (which implies not only that the data boundaries have
not changed, but also that the window and acknowledgment
fields of the header have not changed), then the same IP
Identification field MAY be used (see Section 3.2.1.5).
IMPLEMENTATION:
Some TCP implementors have chosen to "packetize" the
data stream, i.e., to pick segment boundaries when
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segments are originally sent and to queue these
segments in a "retransmission queue" until they are
acknowledged. Another design (which may be simpler) is
to defer packetizing until each time data is
transmitted or retransmitted, so there will be no
segment retransmission queue.
In an implementation with a segment retransmission
queue, TCP performance may be enhanced by repacketizing
the segments awaiting acknowledgment when the first
retransmission timeout occurs. That is, the
outstanding segments that fitted would be combined into
one maximum-sized segment, with a new IP Identification
value. The TCP would then retain this combined segment
in the retransmit queue until it was acknowledged.
However, if the first two segments in the
retransmission queue totalled more than one maximum-
sized segment, the TCP would retransmit only the first
segment using the original IP Identification field.
4.2.2.16 Managing the Window: RFC-793 Section 3.7, page 41
A TCP receiver SHOULD NOT shrink the window, i.e., move the
right window edge to the left. However, a sending TCP MUST
be robust against window shrinking, which may cause the
"useable window" (see Section 4.2.3.4) to become negative.
If this happens, the sender SHOULD NOT send new data, but
SHOULD retransmit normally the old unacknowledged data
between SND.UNA and SND.UNA+SND.WND. The sender MAY also
retransmit old data beyond SND.UNA+SND.WND, but SHOULD NOT
time out the connection if data beyond the right window edge
is not acknowledged. If the window shrinks to zero, the TCP
MUST probe it in the standard way (see next Section).
DISCUSSION:
Many TCP implementations become confused if the window
shrinks from the right after data has been sent into a
larger window. Note that TCP has a heuristic to select
the latest window update despite possible datagram
reordering; as a result, it may ignore a window update
with a smaller window than previously offered if
neither the sequence number nor the acknowledgment
number is increased.
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4.2.2.17 Probing Zero Windows: RFC-793 Section 3.7, page 42
Probing of zero (offered) windows MUST be supported.
A TCP MAY keep its offered receive window closed
indefinitely. As long as the receiving TCP continues to
send acknowledgments in response to the probe segments, the
sending TCP MUST allow the connection to stay open.
DISCUSSION:
It is extremely important to remember that ACK
(acknowledgment) segments that contain no data are not
reliably transmitted by TCP. If zero window probing is
not supported, a connection may hang forever when an
ACK segment that re-opens the window is lost.
The delay in opening a zero window generally occurs
when the receiving application stops taking data from
its TCP. For example, consider a printer daemon
application, stopped because the printer ran out of
paper.
The transmitting host SHOULD send the first zero-window
probe when a zero window has existed for the retransmission
timeout period (see Section 4.2.2.15), and SHOULD increase
exponentially the interval between successive probes.
DISCUSSION:
This procedure minimizes delay if the zero-window
condition is due to a lost ACK segment containing a
window-opening update. Exponential backoff is
recommended, possibly with some maximum interval not
specified here. This procedure is similar to that of
the retransmission algorithm, and it may be possible to
combine the two procedures in the implementation.
4.2.2.18 Passive OPEN Calls: RFC-793 Section 3.8
Every passive OPEN call either creates a new connection
record in LISTEN state, or it returns an error; it MUST NOT
affect any previously created connection record.
A TCP that supports multiple concurrent users MUST provide
an OPEN call that will functionally allow an application to
LISTEN on a port while a connection block with the same
local port is in SYN-SENT or SYN-RECEIVED state.
DISCUSSION:
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Some applications (e.g., SMTP servers) may need to
handle multiple connection attempts at about the same
time. The probability of a connection attempt failing
is reduced by giving the application some means of
listening for a new connection at the same time that an
earlier connection attempt is going through the three-
way handshake.
IMPLEMENTATION:
Acceptable implementations of concurrent opens may
permit multiple passive OPEN calls, or they may allow
"cloning" of LISTEN-state connections from a single
passive OPEN call.
4.2.2.19 Time to Live: RFC-793 Section 3.9, page 52
RFC-793 specified that TCP was to request the IP layer to
send TCP segments with TTL = 60. This is obsolete; the TTL
value used to send TCP segments MUST be configurable. See
Section 3.2.1.7 for discussion.
4.2.2.20 Event Processing: RFC-793 Section 3.9
While it is not strictly required, a TCP SHOULD be capable
of queueing out-of-order TCP segments. Change the "may" in
the last sentence of the first paragraph on page 70 to
"should".
DISCUSSION:
Some small-host implementations have omitted segment
queueing because of limited buffer space. This
omission may be expected to adversely affect TCP
throughput, since loss of a single segment causes all
later segments to appear to be "out of sequence".
In general, the processing of received segments MUST be
implemented to aggregate ACK segments whenever possible.
For example, if the TCP is processing a series of queued
segments, it MUST process them all before sending any ACK
segments.
Here are some detailed error corrections and notes on the
Event Processing section of RFC-793.
(a) CLOSE Call, CLOSE-WAIT state, p. 61: enter LAST-ACK
state, not CLOSING.
(b) LISTEN state, check for SYN (pp. 65, 66): With a SYN
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bit, if the security/compartment or the precedence is
wrong for the segment, a reset is sent. The wrong form
of reset is shown in the text; it should be:
(c) SYN-SENT state, Check for SYN, p. 68: When the
connection enters ESTABLISHED state, the following
variables must be set:
SND.WND
RFC1122 TRANSPORT LAYER -- TCP October 1989
DISCUSSION:
RFC-793 (see page 74) was ambiguous about whether or
not an ACK segment should be sent when an out-of-order
segment was received, i.e., when SEG.SEQ was unequal to
RCV.NXT.
One reason for ACKing out-of-order segments might be to
support an experimental algorithm known as "fast
retransmit". With this algorithm, the sender uses the
"redundant" ACK's to deduce that a segment has been
lost before the retransmission timer has expired. It
counts the number of times an ACK has been received
with the same value of SEG.ACK and with the same right
window edge. If more than a threshold number of such
ACK's is received, then the segment containing the
octets starting at SEG.ACK is assumed to have been lost
and is retransmitted, without awaiting a timeout. The
threshold is chosen to compensate for the maximum
likely segment reordering in the Internet. There is
not yet enough experience with the fast retransmit
algorithm to determine how useful it is.
4.2.3 SPECIFIC ISSUES
4.2.3.1 Retransmission Timeout Calculation
A host TCP MUST implement Karn's algorithm and Jacobson's
algorithm for computing the retransmission timeout ("RTO").
o Jacobson's algorithm for computing the smoothed round-
trip ("RTT") time incorporates a simple measure of the
variance [TCP:7].
o Karn's algorithm for selecting RTT measurements ensures
that ambiguous round-trip times will not corrupt the
calculation of the smoothed round-trip time [TCP:6].
This implementation also MUST include "exponential backoff"
for successive RTO values for the same segment.
Retransmission of SYN segments SHOULD use the same algorithm
as data segments.
DISCUSSION:
There were two known problems with the RTO calculations
specified in RFC-793. First, the accurate measurement
of RTTs is difficult when there are retransmissions.
Second, the algorithm to compute the smoothed round-
trip time is inadequate [TCP:7], because it incorrectly
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assumed that the variance in RTT values would be small
and constant. These problems were solved by Karn's and
Jacobson's algorithm, respectively.
The performance increase resulting from the use of
these improvements varies from noticeable to dramatic.
Jacobson's algorithm for incorporating the measured RTT
variance is especially important on a low-speed link,
where the natural variation of packet sizes causes a
large variation in RTT. One vendor found link
utilization on a 9.6kb line went from 10% to 90% as a
result of implementing Jacobson's variance algorithm in
TCP.
The following values SHOULD be used to initialize the
estimation parameters for a new connection:
(a) RTT = 0 seconds.
(b) RTO = 3 seconds. (The smoothed variance is to be
initialized to the value that will result in this RTO).
The recommended upper and lower bounds on the RTO are known
to be inadequate on large internets. The lower bound SHOULD
be measured in fractions of a second (to accommodate high
speed LANs) and the upper bound should be 2*MSL, i.e., 240
seconds.
DISCUSSION:
Experience has shown that these initialization values
are reasonable, and that in any case the Karn and
Jacobson algorithms make TCP behavior reasonably
insensitive to the initial parameter choices.
4.2.3.2 When to Send an ACK Segment
A host that is receiving a stream of TCP data segments can
increase efficiency in both the Internet and the hosts by
sending fewer than one ACK (acknowledgment) segment per data
segment received; this is known as a "delayed ACK" [TCP:5].
A TCP SHOULD implement a delayed ACK, but an ACK should not
be excessively delayed; in particular, the delay MUST be
less than 0.5 seconds, and in a stream of full-sized
segments there SHOULD be an ACK for at least every second
segment.
DISCUSSION:
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A delayed ACK gives the application an opportunity to
update the window and perhaps to send an immediate
response. In particular, in the case of character-mode
remote login, a delayed ACK can reduce the number of
segments sent by the server by a factor of 3 (ACK,
window update, and echo character all combined in one
segment).
In addition, on some large multi-user hosts, a delayed
ACK can substantially reduce protocol processing
overhead by reducing the total number of packets to be
processed [TCP:5]. However, excessive delays on ACK's
can disturb the round-trip timing and packet "clocking"
algorithms [TCP:7].
4.2.3.3 When to Send a Window Update
A TCP MUST include a SWS avoidance algorithm in the receiver
[TCP:5].
IMPLEMENTATION:
The receiver's SWS avoidance algorithm determines when
the right window edge may be advanced; this is
customarily known as "updating the window". This
algorithm combines with the delayed ACK algorithm (see
Section 4.2.3.2) to determine when an ACK segment
containing the current window will really be sent to
the receiver. We use the notation of RFC-793; see
Figures 4 and 5 in that document.
The solution to receiver SWS is to avoid advancing the
right window edge RCV.NXT+RCV.WND in small increments,
even if data is received from the network in small
segments.
Suppose the total receive buffer space is RCV.BUFF. At
any given moment, RCV.USER octets of this total may be
tied up with data that has been received and
acknowledged but which the user process has not yet
consumed. When the connection is quiescent, RCV.WND =
RCV.BUFF and RCV.USER = 0.
Keeping the right window edge fixed as data arrives and
is acknowledged requires that the receiver offer less
than its full buffer space, i.e., the receiver must
specify a RCV.WND that keeps RCV.NXT+RCV.WND constant
as RCV.NXT increases. Thus, the total buffer space
RCV.BUFF is generally divided into three parts:
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||
1 2 3
----|---------|------------------|------|----
RCV.NXT ^
(Fixed)
1 - RCV.USER = data received but not yet consumed;
2 - RCV.WND = space advertised to sender;
3 - Reduction = space available but not yet
advertised.
The suggested SWS avoidance algorithm for the receiver
is to keep RCV.NXT+RCV.WND fixed until the reduction
satisfies:
RCV.BUFF - RCV.USER - RCV.WND >=
min( Fr * RCV.BUFF, Eff.snd.MSS )
where Fr is a fraction whose recommended value is 1/2,
and Eff.snd.MSS is the effective send MSS for the
connection (see Section 4.2.2.6). When the inequality
is satisfied, RCV.WND is set to RCV.BUFF-RCV.USER.
Note that the general effect of this algorithm is to
advance RCV.WND in increments of Eff.snd.MSS (for
realistic receive buffers: Eff.snd.MSS < RCV.BUFF/2).
Note also that the receiver must use its own
Eff.snd.MSS, assuming it is the same as the sender's.
4.2.3.4 When to Send Data
A TCP MUST include a SWS avoidance algorithm in the sender.
A TCP SHOULD implement the Nagle Algorithm [TCP:9] to
coalesce short segments. However, there MUST be a way for
an application to disable the Nagle algorithm on an
individual connection. In all cases, sending data is also
subject to the limitation imposed by the Slow Start
algorithm (Section 4.2.2.15).
DISCUSSION:
The Nagle algorithm is generally as follows:
If there is unacknowledged data (i.e., SND.NXT >
SND.UNA), then the sending TCP buffers all user
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data (regardless of the PSH bit), until the
outstanding data has been acknowledged or until
the TCP can send a full-sized segment (Eff.snd.MSS
bytes; see Section 4.2.2.6).
Some applications (e.g., real-time display window
updates) require that the Nagle algorithm be turned
off, so small data segments can be streamed out at the
maximum rate.
IMPLEMENTATION:
The sender's SWS avoidance algorithm is more difficult
than the receivers's, because the sender does not know
(directly) the receiver's total buffer space RCV.BUFF.
An approach which has been found to work well is for
the sender to calculate Max(SND.WND), the maximum send
window it has seen so far on the connection, and to use
this value as an estimate of RCV.BUFF. Unfortunately,
this can only be an estimate; the receiver may at any
time reduce the size of RCV.BUFF. To avoid a resulting
deadlock, it is necessary to have a timeout to force
transmission of data, overriding the SWS avoidance
algorithm. In practice, this timeout should seldom
occur.
The "useable window" [TCP:5] is:
U = SND.UNA + SND.WND - SND.NXT
i.e., the offered window less the amount of data sent
but not acknowledged. If D is the amount of data
queued in the sending TCP but not yet sent, then the
following set of rules is recommended.
Send data:
(1) if a maximum-sized segment can be sent, i.e, if:
min(D,U) >= Eff.snd.MSS;
(2) or if the data is pushed and all queued data can
be sent now, i.e., if:
[SND.NXT = SND.UNA and] PUSHED and D <= U
(the bracketed condition is imposed by the Nagle
algorithm);
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(3) or if at least a fraction Fs of the maximum window
can be sent, i.e., if:
[SND.NXT = SND.UNA and]
min(D.U) >= Fs * Max(SND.WND);
(4) or if data is PUSHed and the override timeout
occurs.
Here Fs is a fraction whose recommended value is 1/2.
The override timeout should be in the range 0.1 - 1.0
seconds. It may be convenient to combine this timer
with the timer used to probe zero windows (Section
4.2.2.17).
Finally, note that the SWS avoidance algorithm just
specified is to be used instead of the sender-side
algorithm contained in [TCP:5].
4.2.3.5 TCP Connection Failures
Excessive retransmission of the same segment by TCP
indicates some failure of the remote host or the Internet
path. This failure may be of short or long duration. The
following procedure MUST be used to handle excessive
retransmissions of data segments [IP:11]:
(a) There are two thresholds R1 and R2 measuring the amount
of retransmission that has occurred for the same
segment. R1 and R2 might be measured in time units or
as a count of retransmissions.
(b) When the number of transmissions of the same segment
reaches or exceeds threshold R1, pass negative advice
(see Section 3.3.1.4) to the IP layer, to trigger
dead-gateway diagnosis.
(c) When the number of transmissions of the same segment
reaches a threshold R2 greater than R1, close the
connection.
(d) An application MUST be able to set the value for R2 for
a particular connection. For example, an interactive
application might set R2 to "infinity," giving the user
control over when to disconnect.
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(d) TCP SHOULD inform the application of the delivery
problem (unless such information has been disabled by
the application; see Section 4.2.4.1), when R1 is
reached and before R2. This will allow a remote login
(User Telnet) application program to inform the user,
for example.
The value of R1 SHOULD correspond to at least 3
retransmissions, at the current RTO. The value of R2 SHOULD
correspond to at least 100 seconds.
An attempt to open a TCP connection could fail with
excessive retransmissions of the SYN segment or by receipt
of a RST segment or an ICMP Port Unreachable. SYN
retransmissions MUST be handled in the general way just
described for data retransmissions, including notification
of the application layer.
However, the values of R1 and R2 may be different for SYN
and data segments. In particular, R2 for a SYN segment MUST
be set large enough to provide retransmission of the segment
for at least 3 minutes. The application can close the
connection (i.e., give up on the open attempt) sooner, of
course.
DISCUSSION:
Some Internet paths have significant setup times, and
the number of such paths is likely to increase in the
future.
4.2.3.6 TCP Keep-Alives
Implementors MAY include "keep-alives" in their TCP
implementations, although this practice is not universally
accepted. If keep-alives are included, the application MUST
be able to turn them on or off for each TCP connection, and
they MUST default to off.
Keep-alive packets MUST only be sent when no data or
acknowledgement packets have been received for the
connection within an interval. This interval MUST be
configurable and MUST default to no less than two hours.
It is extremely important to remember that ACK segments that
contain no data are not reliably transmitted by TCP.
Consequently, if a keep-alive mechanism is implemented it
MUST NOT interpret failure to respond to any specific probe
as a dead connection.
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An implementation SHOULD send a keep-alive segment with no
data; however, it MAY be configurable to send a keep-alive
segment containing one garbage octet, for compatibility with
erroneous TCP implementations.
DISCUSSION:
A "keep-alive" mechanism periodically probes the other
end of a connection when the connection is otherwise
idle, even when there is no data to be sent. The TCP
specification does not include a keep-alive mechanism
because it could: (1) cause perfectly good connections
to break during transient Internet failures; (2)
consume unnecessary bandwidth ("if no one is using the
connection, who cares if it is still good?"); and (3)
cost money for an Internet path that charges for
packets.
Some TCP implementations, however, have included a
keep-alive mechanism. To confirm that an idle
connection is still active, these implementations send
a probe segment designed to elicit a response from the
peer TCP. Such a segment generally contains SEG.SEQ =
SND.NXT-1 and may or may not contain one garbage octet
of data. Note that on a quiet connection SND.NXT =
RCV.NXT, so that this SEG.SEQ will be outside the
window. Therefore, the probe causes the receiver to
return an acknowledgment segment, confirming that the
connection is still live. If the peer has dropped the
connection due to a network partition or a crash, it
will respond with a RST instead of an acknowledgment
segment.
Unfortunately, some misbehaved TCP implementations fail
to respond to a segment with SEG.SEQ = SND.NXT-1 unless
the segment contains data. Alternatively, an
implementation could determine whether a peer responded
correctly to keep-alive packets with no garbage data
octet.
A TCP keep-alive mechanism should only be invoked in
server applications that might otherwise hang
indefinitely and consume resources unnecessarily if a
client crashes or aborts a connection during a network
failure.
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4.2.3.7 TCP Multihoming
If an application on a multihomed host does not specify the
local IP address when actively opening a TCP connection,
then the TCP MUST ask the IP layer to select a local IP
address before sending the (first) SYN. See the function
GET_SRCADDR() in Section 3.4.
At all other times, a previous segment has either been sent
or received on this connection, and TCP MUST use the same
local address is used that was used in those previous
segments.
4.2.3.8 IP Options
When received options are passed up to TCP from the IP
layer, TCP MUST ignore options that it does not understand.
A TCP MAY support the Time Stamp and Record Route options.
An application MUST be able to specify a source route when
it actively opens a TCP connection, and this MUST take
precedence over a source route received in a datagram.
When a TCP connection is OPENed passively and a packet
arrives with a completed IP Source Route option (containing
a return route), TCP MUST save the return route and use it
for all segments sent on this connection. If a different
source route arrives in a later segment, the later
definition SHOULD override the earlier one.
4.2.3.9 ICMP Messages
TCP MUST act on an ICMP error message passed up from the IP
layer, directing it to the connection that created the
error. The necessary demultiplexing information can be
found in the IP header contained within the ICMP message.
o Source Quench
TCP MUST react to a Source Quench by slowing
transmission on the connection. The RECOMMENDED
procedure is for a Source Quench to trigger a "slow
start," as if a retransmission timeout had occurred.
o Destination Unreachable -- codes 0, 1, 5
Since these Unreachable messages indicate soft error
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conditions, TCP MUST NOT abort the connection, and it
SHOULD make the information available to the
application.
DISCUSSION:
TCP could report the soft error condition directly
to the application layer with an upcall to the
ERROR_REPORT routine, or it could merely note the
message and report it to the application only when
and if the TCP connection times out.
o Destination Unreachable -- codes 2-4
These are hard error conditions, so TCP SHOULD abort
the connection.
o Time Exceeded -- codes 0, 1
This should be handled the same way as Destination
Unreachable codes 0, 1, 5 (see above).
o Parameter Problem
This should be handled the same way as Destination
Unreachable codes 0, 1, 5 (see above).
4.2.3.10 Remote Address Validation
A TCP implementation MUST reject as an error a local OPEN
call for an invalid remote IP address (e.g., a broadcast or
multicast address).
An incoming SYN with an invalid source address must be
ignored either by TCP or by the IP layer (see Section
3.2.1.3).
A TCP implementation MUST silently discard an incoming SYN
segment that is addressed to a broadcast or multicast
address.
4.2.3.11 TCP Traffic Patterns
IMPLEMENTATION:
The TCP protocol specification [TCP:1] gives the
implementor much freedom in designing the algorithms
that control the message flow over the connection --
packetizing, managing the window, sending
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acknowledgments, etc. These design decisions are
difficult because a TCP must adapt to a wide range of
traffic patterns. Experience has shown that a TCP
implementor needs to verify the design on two extreme
traffic patterns:
o Single-character Segments
Even if the sender is using the Nagle Algorithm,
when a TCP connection carries remote login traffic
across a low-delay LAN the receiver will generally
get a stream of single-character segments. If
remote terminal echo mode is in effect, the
receiver's system will generally echo each
character as it is received.
o Bulk Transfer
When TCP is used for bulk transfer, the data
stream should be made up (almost) entirely of
segments of the size of the effective MSS.
Although TCP uses a sequence number space with
byte (octet) granularity, in bulk-transfer mode
its operation should be as if TCP used a sequence
space that counted only segments.
Experience has furthermore shown that a single TCP can
effectively and efficiently handle these two extremes.
The most important tool for verifying a new TCP
implementation is a packet trace program. There is a
large volume of experience showing the importance of
tracing a variety of traffic patterns with other TCP
implementations and studying the results carefully.
4.2.3.12 Efficiency
IMPLEMENTATION:
Extensive experience has led to the following
suggestions for efficient implementation of TCP:
(a) Don't Copy Data
In bulk data transfer, the primary CPU-intensive
tasks are copying data from one place to another
and checksumming the data. It is vital to
minimize the number of copies of TCP data. Since
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the ultimate speed limitation may be fetching data
across the memory bus, it may be useful to combine
the copy with checksumming, doing both with a
single memory fetch.
(b) Hand-Craft the Checksum Routine
A good TCP checksumming routine is typically two
to five times faster than a simple and direct
implementation of the definition. Great care and
clever coding are often required and advisable to
make the checksumming code "blazing fast". See
[TCP:10].
(c) Code for the Common Case
TCP protocol processing can be complicated, but
for most segments there are only a few simple
decisions to be made. Per-segment processing will
be greatly speeded up by coding the main line to
minimize the number of decisions in the most
common case.
4.2.4 TCP/APPLICATION LAYER INTERFACE
4.2.4.1 Asynchronous Reports
There MUST be a mechanism for reporting soft TCP error
conditions to the application. Generically, we assume this
takes the form of an application-supplied ERROR_REPORT
routine that may be upcalled [INTRO:7] asynchronously from
the transport layer:
ERROR_REPORT(local connection name, reason, subreason)
The precise encoding of the reason and subreason parameters
is not specified here. However, the conditions that are
reported asynchronously to the application MUST include:
* ICMP error message arrived (see 4.2.3.9)
* Excessive retransmissions (see 4.2.3.5)
* Urgent pointer advance (see 4.2.2.4).
However, an application program that does not want to
receive such ERROR_REPORT calls SHOULD be able to
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effectively disable these calls.
DISCUSSION:
These error reports generally reflect soft errors that
can be ignored without harm by many applications. It
has been suggested that these error report calls should
default to "disabled," but this is not required.
4.2.4.2 Type-of-Service
The application layer MUST be able to specify the Type-of-
Service (TOS) for segments that are sent on a connection.
It not required, but the application SHOULD be able to
change the TOS during the connection lifetime. TCP SHOULD
pass the current TOS value without change to the IP layer,
when it sends segments on the connection.
The TOS will be specified independently in each direction on
the connection, so that the receiver application will
specify the TOS used for ACK segments.
TCP MAY pass the most recently received TOS up to the
application.
DISCUSSION
Some applications (e.g., SMTP) change the nature of
their communication during the lifetime of a
connection, and therefore would like to change the TOS
specification.
Note also that the OPEN call specified in RFC-793
includes a parameter ("options") in which the caller
can specify IP options such as source route, record
route, or timestamp.
4.2.4.3 Flush Call
Some TCP implementations have included a FLUSH call, which
will empty the TCP send queue of any data for which the user
has issued SEND calls but which is still to the right of the
current send window. That is, it flushes as much queued
send data as possible without losing sequence number
synchronization. This is useful for implementing the "abort
output" function of Telnet.
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4.2.4.4 Multihoming
The user interface outlined in sections 2.7 and 3.8 of RFC-
793 needs to be extended for multihoming. The OPEN call
MUST have an optional parameter:
OPEN( ... [local IP address,] ... )
to allow the specification of the local IP address.
DISCUSSION:
Some TCP-based applications need to specify the local
IP address to be used to open a particular connection;
FTP is an example.
IMPLEMENTATION:
A passive OPEN call with a specified "local IP address"
parameter will await an incoming connection request to
that address. If the parameter is unspecified, a
passive OPEN will await an incoming connection request
to any local IP address, and then bind the local IP
address of the connection to the particular address
that is used.
For an active OPEN call, a specified "local IP address"
parameter will be used for opening the connection. If
the parameter is unspecified, the networking software
will choose an appropriate local IP address (see
Section 3.3.4.2) for the connection
4.2.5 TCP REQUIREMENT SUMMARY
| | | | |S| |
| | | | |H| |F
| | | | |O|M|o
| | |S| |U|U|o
| | |H| |L|S|t
| |M|O| |D|T|n
| |U|U|M| | |o
| |S|L|A|N|N|t
| |T|D|Y|O|O|t
FEATURE |SECTION | | | |T|T|e
-------------------------------------------------|--------|-|-|-|-|-|--
| | | | | | |
Push flag | | | | | | |
Aggregate or queue un-pushed data |4.2.2.2 | | |x| | |
Sender collapse successive PSH flags |4.2.2.2 | |x| | | |
SEND call can specify PUSH |4.2.2.2 | | |x| | |
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If cannot: sender buffer indefinitely |4.2.2.2 | | | | |x|
If cannot: PSH last segment |4.2.2.2 |x| | | | |
Notify receiving ALP of PSH |4.2.2.2 | | |x| | |1
Send max size segment when possible |4.2.2.2 | |x| | | |
| | | | | | |
Window | | | | | | |
Treat as unsigned number |4.2.2.3 |x| | | | |
Handle as 32-bit number |4.2.2.3 | |x| | | |
Shrink window from right |4.2.2.16| | | |x| |
Robust against shrinking window |4.2.2.16|x| | | | |
Receiver's window closed indefinitely |4.2.2.17| | |x| | |
Sender probe zero window |4.2.2.17|x| | | | |
First probe after RTO |4.2.2.17| |x| | | |
Exponential backoff |4.2.2.17| |x| | | |
Allow window stay zero indefinitely |4.2.2.17|x| | | | |
Sender timeout OK conn with zero wind |4.2.2.17| | | | |x|
| | | | | | |
Urgent Data | | | | | | |
Pointer points to last octet |4.2.2.4 |x| | | | |
Arbitrary length urgent data sequence |4.2.2.4 |x| | | | |
Inform ALP asynchronously of urgent data |4.2.2.4 |x| | | | |1
ALP can learn if/how much urgent data Q'd |4.2.2.4 |x| | | | |1
| | | | | | |
TCP Options | | | | | | |
Receive TCP option in any segment |4.2.2.5 |x| | | | |
Ignore unsupported options |4.2.2.5 |x| | | | |
Cope with illegal option length |4.2.2.5 |x| | | | |
Implement sending & receiving MSS option |4.2.2.6 |x| | | | |
Send MSS option unless 536 |4.2.2.6 | |x| | | |
Send MSS option always |4.2.2.6 | | |x| | |
Send-MSS default is 536 |4.2.2.6 |x| | | | |
Calculate effective send seg size |4.2.2.6 |x| | | | |
| | | | | | |
TCP Checksums | | | | | | |
Sender compute checksum |4.2.2.7 |x| | | | |
Receiver check checksum |4.2.2.7 |x| | | | |
| | | | | | |
Use clock-driven ISN selection |4.2.2.9 |x| | | | |
| | | | | | |
Opening Connections | | | | | | |
Support simultaneous open attempts |4.2.2.10|x| | | | |
SYN-RCVD remembers last state |4.2.2.11|x| | | | |
Passive Open call interfere with others |4.2.2.18| | | | |x|
Function: simultan. LISTENs for same port |4.2.2.18|x| | | | |
Ask IP for src address for SYN if necc. |4.2.3.7 |x| | | | |
Otherwise, use local addr of conn. |4.2.3.7 |x| | | | |
OPEN to broadcast/multicast IP Address |4.2.3.14| | | | |x|
Silently discard seg to bcast/mcast addr |4.2.3.14|x| | | | |
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| | | | | | |
Closing Connections | | | | | | |
RST can contain data |4.2.2.12| |x| | | |
Inform application of aborted conn |4.2.2.13|x| | | | |
Half-duplex close connections |4.2.2.13| | |x| | |
Send RST to indicate data lost |4.2.2.13| |x| | | |
In TIME-WAIT state for 2xMSL seconds |4.2.2.13|x| | | | |
Accept SYN from TIME-WAIT state |4.2.2.13| | |x| | |
| | | | | | |
Retransmissions | | | | | | |
Jacobson Slow Start algorithm |4.2.2.15|x| | | | |
Jacobson Congestion-Avoidance algorithm |4.2.2.15|x| | | | |
Retransmit with same IP ident |4.2.2.15| | |x| | |
Karn's algorithm |4.2.3.1 |x| | | | |
Jacobson's RTO estimation alg. |4.2.3.1 |x| | | | |
Exponential backoff |4.2.3.1 |x| | | | |
SYN RTO calc same as data |4.2.3.1 | |x| | | |
Recommended initial values and bounds |4.2.3.1 | |x| | | |
| | | | | | |
Generating ACK's: | | | | | | |
Queue out-of-order segments |4.2.2.20| |x| | | |
Process all Q'd before send ACK |4.2.2.20|x| | | | |
Send ACK for out-of-order segment |4.2.2.21| | |x| | |
Delayed ACK's |4.2.3.2 | |x| | | |
Delay < 0.5 seconds |4.2.3.2 |x| | | | |
Every 2nd full-sized segment ACK'd |4.2.3.2 |x| | | | |
Receiver SWS-Avoidance Algorithm |4.2.3.3 |x| | | | |
| | | | | | |
Sending data | | | | | | |
Configurable TTL |4.2.2.19|x| | | | |
Sender SWS-Avoidance Algorithm |4.2.3.4 |x| | | | |
Nagle algorithm |4.2.3.4 | |x| | | |
Application can disable Nagle algorithm |4.2.3.4 |x| | | | |
| | | | | | |
Connection Failures: | | | | | | |
Negative advice to IP on R1 retxs |4.2.3.5 |x| | | | |
Close connection on R2 retxs |4.2.3.5 |x| | | | |
ALP can set R2 |4.2.3.5 |x| | | | |1
Inform ALP of R1<=retxs